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authorMatt A. Tobin <mattatobin@localhost.localdomain>2018-02-02 04:16:08 -0500
committerMatt A. Tobin <mattatobin@localhost.localdomain>2018-02-02 04:16:08 -0500
commit5f8de423f190bbb79a62f804151bc24824fa32d8 (patch)
tree10027f336435511475e392454359edea8e25895d /dom/media/AudioStream.h
parent49ee0794b5d912db1f95dce6eb52d781dc210db5 (diff)
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Add m-esr52 at 52.6.0
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+/* -*- Mode: C++; tab-width: 2; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
+/* vim:set ts=2 sw=2 sts=2 et cindent: */
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this
+ * file, You can obtain one at http://mozilla.org/MPL/2.0/. */
+#if !defined(AudioStream_h_)
+#define AudioStream_h_
+
+#include "AudioSampleFormat.h"
+#include "nsAutoPtr.h"
+#include "nsCOMPtr.h"
+#include "nsThreadUtils.h"
+#include "mozilla/dom/AudioChannelBinding.h"
+#include "mozilla/Monitor.h"
+#include "mozilla/RefPtr.h"
+#include "mozilla/TimeStamp.h"
+#include "mozilla/UniquePtr.h"
+#include "CubebUtils.h"
+#include "soundtouch/SoundTouchFactory.h"
+
+namespace mozilla {
+
+struct CubebDestroyPolicy
+{
+ void operator()(cubeb_stream* aStream) const {
+ cubeb_stream_destroy(aStream);
+ }
+};
+
+class AudioStream;
+class FrameHistory;
+class AudioConfig;
+class AudioConverter;
+
+class AudioClock
+{
+public:
+ AudioClock();
+
+ // Initialize the clock with the current sampling rate.
+ // Need to be called before querying the clock.
+ void Init(uint32_t aRate);
+
+ // 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);
+
+ /**
+ * @param aFrames The playback position in frames of the audio engine.
+ * @return The playback position in frames of the stream,
+ * adjusted by playback rate changes and underrun frames.
+ */
+ int64_t GetPositionInFrames(int64_t aFrames) const;
+
+ /**
+ * @param frames The playback position in frames of the audio engine.
+ * @return The playback position in microseconds of the stream,
+ * adjusted by playback rate changes and underrun frames.
+ */
+ int64_t GetPosition(int64_t frames) const;
+
+ // Set the playback rate.
+ // Called on the audio thread.
+ void SetPlaybackRate(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;
+
+ uint32_t GetInputRate() const { return mInRate; }
+ uint32_t GetOutputRate() const { return mOutRate; }
+
+private:
+ // Output rate in Hz (characteristic of the playback rate)
+ uint32_t mOutRate;
+ // Input rate in Hz (characteristic of the media being played)
+ uint32_t 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<FrameHistory> mFrameHistory;
+};
+
+/*
+ * A bookkeeping class to track the read/write position of an audio buffer.
+ */
+class AudioBufferCursor {
+public:
+ AudioBufferCursor(AudioDataValue* aPtr, uint32_t aChannels, uint32_t aFrames)
+ : mPtr(aPtr), mChannels(aChannels), mFrames(aFrames) {}
+
+ // Advance the cursor to account for frames that are consumed.
+ uint32_t Advance(uint32_t aFrames) {
+ MOZ_ASSERT(mFrames >= aFrames);
+ mFrames -= aFrames;
+ mPtr += mChannels * aFrames;
+ return aFrames;
+ }
+
+ // The number of frames available for read/write in this buffer.
+ uint32_t Available() const { return mFrames; }
+
+ // Return a pointer where read/write should begin.
+ AudioDataValue* Ptr() const { return mPtr; }
+
+protected:
+ AudioDataValue* mPtr;
+ const uint32_t mChannels;
+ uint32_t mFrames;
+};
+
+/*
+ * A helper class to encapsulate pointer arithmetic and provide means to modify
+ * the underlying audio buffer.
+ */
+class AudioBufferWriter : private AudioBufferCursor {
+public:
+ AudioBufferWriter(AudioDataValue* aPtr, uint32_t aChannels, uint32_t aFrames)
+ : AudioBufferCursor(aPtr, aChannels, aFrames) {}
+
+ uint32_t WriteZeros(uint32_t aFrames) {
+ memset(mPtr, 0, sizeof(AudioDataValue) * mChannels * aFrames);
+ return Advance(aFrames);
+ }
+
+ uint32_t Write(const AudioDataValue* aPtr, uint32_t aFrames) {
+ memcpy(mPtr, aPtr, sizeof(AudioDataValue) * mChannels * aFrames);
+ return Advance(aFrames);
+ }
+
+ // Provide a write fuction to update the audio buffer with the following
+ // signature: uint32_t(const AudioDataValue* aPtr, uint32_t aFrames)
+ // aPtr: Pointer to the audio buffer.
+ // aFrames: The number of frames available in the buffer.
+ // return: The number of frames actually written by the function.
+ template <typename Function>
+ uint32_t Write(const Function& aFunction, uint32_t aFrames) {
+ return Advance(aFunction(mPtr, aFrames));
+ }
+
+ using AudioBufferCursor::Available;
+};
+
+// 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},
+// SetMicrophoneActive is thread-safe without external synchronization.
+class AudioStream final
+{
+ virtual ~AudioStream();
+
+public:
+ NS_INLINE_DECL_THREADSAFE_REFCOUNTING(AudioStream)
+
+ class Chunk {
+ public:
+ // Return a pointer to the audio data.
+ virtual const AudioDataValue* Data() const = 0;
+ // Return the number of frames in this chunk.
+ virtual uint32_t Frames() const = 0;
+ // Return the number of audio channels.
+ virtual uint32_t Channels() const = 0;
+ // Return the sample rate of this chunk.
+ virtual uint32_t Rate() const = 0;
+ // Return a writable pointer for downmixing.
+ virtual AudioDataValue* GetWritable() const = 0;
+ virtual ~Chunk() {}
+ };
+
+ class DataSource {
+ public:
+ // Return a chunk which contains at most aFrames frames or zero if no
+ // frames in the source at all.
+ virtual UniquePtr<Chunk> PopFrames(uint32_t aFrames) = 0;
+ // Return true if no more data will be added to the source.
+ virtual bool Ended() const = 0;
+ // Notify that all data is drained by the AudioStream.
+ virtual void Drained() = 0;
+ protected:
+ virtual ~DataSource() {}
+ };
+
+ explicit AudioStream(DataSource& aSource);
+
+ // 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(uint32_t aNumChannels, uint32_t aRate,
+ const dom::AudioChannel aAudioStreamChannel);
+
+ // Closes the stream. All future use of the stream is an error.
+ void Shutdown();
+
+ void Reset();
+
+ // 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);
+
+ // Start the stream.
+ void Start();
+
+ // 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();
+
+ static uint32_t GetPreferredRate()
+ {
+ return CubebUtils::PreferredSampleRate();
+ }
+
+ uint32_t 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:
+ nsresult OpenCubeb(cubeb* aContext, cubeb_stream_params& aParams,
+ TimeStamp aStartTime, bool aIsFirst);
+
+ static long DataCallback_S(cubeb_stream*, void* aThis,
+ const void* /* aInputBuffer */, void* aOutputBuffer,
+ long aFrames)
+ {
+ return static_cast<AudioStream*>(aThis)->DataCallback(aOutputBuffer, aFrames);
+ }
+
+ static void StateCallback_S(cubeb_stream*, void* aThis, cubeb_state aState)
+ {
+ static_cast<AudioStream*>(aThis)->StateCallback(aState);
+ }
+
+
+ long DataCallback(void* aBuffer, long aFrames);
+ void StateCallback(cubeb_state aState);
+
+ nsresult EnsureTimeStretcherInitializedUnlocked();
+
+ // Return true if audio frames are valid (correct sampling rate and valid
+ // channel count) otherwise false.
+ bool IsValidAudioFormat(Chunk* aChunk);
+
+ void GetUnprocessed(AudioBufferWriter& aWriter);
+ void GetTimeStretched(AudioBufferWriter& aWriter);
+
+ template <typename Function, typename... Args>
+ int InvokeCubeb(Function aFunction, Args&&... aArgs);
+
+ // The monitor is held to protect all access to member variables.
+ Monitor mMonitor;
+
+ uint32_t mChannels;
+ uint32_t mOutChannels;
+ AudioClock mAudioClock;
+ soundtouch::SoundTouch* mTimeStretcher;
+
+ // Output file for dumping audio
+ FILE* mDumpFile;
+
+ // Owning reference to a cubeb_stream.
+ UniquePtr<cubeb_stream, CubebDestroyPolicy> mCubebStream;
+
+ enum StreamState {
+ INITIALIZED, // Initialized, playback has not begun.
+ STARTED, // cubeb started.
+ STOPPED, // Stopped by a call to Pause().
+ DRAINED, // StateCallback has indicated that the drain is complete.
+ ERRORED, // Stream disabled due to an internal error.
+ SHUTDOWN // Shutdown has been called
+ };
+
+ StreamState mState;
+
+ DataSource& mDataSource;
+};
+
+} // namespace mozilla
+
+#endif