mirror of
https://github.com/mozilla/gecko-dev.git
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480 lines
16 KiB
C++
480 lines
16 KiB
C++
/* -*- Mode: C++; tab-width: 2; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
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/* vim:set ts=2 sw=2 sts=2 et cindent: */
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/* This Source Code Form is subject to the terms of the Mozilla Public
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* License, v. 2.0. If a copy of the MPL was not distributed with this
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* file, You can obtain one at http://mozilla.org/MPL/2.0/. */
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#if !defined(AudioStream_h_)
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#define AudioStream_h_
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#include "AudioSampleFormat.h"
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#include "nsAutoPtr.h"
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#include "nsAutoRef.h"
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#include "nsCOMPtr.h"
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#include "nsThreadUtils.h"
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#include "Latency.h"
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#include "mozilla/dom/AudioChannelBinding.h"
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#include "mozilla/StaticMutex.h"
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#include "mozilla/RefPtr.h"
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#include "cubeb/cubeb.h"
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template <>
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class nsAutoRefTraits<cubeb_stream> : public nsPointerRefTraits<cubeb_stream>
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{
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public:
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static void Release(cubeb_stream* aStream) { cubeb_stream_destroy(aStream); }
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};
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namespace soundtouch {
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class SoundTouch;
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}
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namespace mozilla {
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class AudioStream;
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class AudioClock
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{
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public:
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AudioClock(AudioStream* aStream);
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// Initialize the clock with the current AudioStream. Need to be called
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// before querying the clock. Called on the audio thread.
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void Init();
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// Update the number of samples that has been written in the audio backend.
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// Called on the state machine thread.
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void UpdateWritePosition(uint32_t aCount);
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// Get the read position of the stream, in microseconds.
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// Called on the state machine thead.
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// Assumes the AudioStream lock is held and thus calls Unlocked versions
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// of AudioStream funcs.
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uint64_t GetPositionUnlocked();
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// Get the read position of the stream, in frames.
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// Called on the state machine thead.
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uint64_t GetPositionInFrames();
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// Set the playback rate.
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// Called on the audio thread.
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// Assumes the AudioStream lock is held and thus calls Unlocked versions
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// of AudioStream funcs.
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void SetPlaybackRateUnlocked(double aPlaybackRate);
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// Get the current playback rate.
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// Called on the audio thread.
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double GetPlaybackRate();
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// Set if we are preserving the pitch.
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// Called on the audio thread.
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void SetPreservesPitch(bool aPreservesPitch);
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// Get the current pitch preservation state.
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// Called on the audio thread.
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bool GetPreservesPitch();
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private:
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// This AudioStream holds a strong reference to this AudioClock. This
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// pointer is garanteed to always be valid.
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AudioStream* mAudioStream;
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// The old output rate, to compensate audio latency for the period inbetween
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// the moment resampled buffers are pushed to the hardware and the moment the
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// clock should take the new rate into account for A/V sync.
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int mOldOutRate;
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// Position at which the last playback rate change occured
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int64_t mBasePosition;
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// Offset, in frames, at which the last playback rate change occured
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int64_t mBaseOffset;
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// Old base offset (number of samples), used when changing rate to compute the
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// position in the stream.
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int64_t mOldBaseOffset;
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// Old base position (number of microseconds), when changing rate. This is the
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// time in the media, not wall clock position.
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int64_t mOldBasePosition;
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// Write position at which the playbackRate change occured.
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int64_t mPlaybackRateChangeOffset;
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// The previous position reached in the media, used when compensating
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// latency, to have the position at which the playbackRate change occured.
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int64_t mPreviousPosition;
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// Number of samples effectivelly written in backend, i.e. write position.
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int64_t mWritten;
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// Output rate in Hz (characteristic of the playback rate)
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int mOutRate;
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// Input rate in Hz (characteristic of the media being played)
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int mInRate;
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// True if the we are timestretching, false if we are resampling.
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bool mPreservesPitch;
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// True if we are playing at the old playbackRate after it has been changed.
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bool mCompensatingLatency;
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};
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class CircularByteBuffer
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{
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public:
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CircularByteBuffer()
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: mBuffer(nullptr), mCapacity(0), mStart(0), mCount(0)
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{}
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// Set the capacity of the buffer in bytes. Must be called before any
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// call to append or pop elements.
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void SetCapacity(uint32_t aCapacity) {
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NS_ABORT_IF_FALSE(!mBuffer, "Buffer allocated.");
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mCapacity = aCapacity;
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mBuffer = new uint8_t[mCapacity];
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}
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uint32_t Length() {
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return mCount;
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}
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uint32_t Capacity() {
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return mCapacity;
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}
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uint32_t Available() {
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return Capacity() - Length();
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}
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// Append aLength bytes from aSrc to the buffer. Caller must check that
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// sufficient space is available.
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void AppendElements(const uint8_t* aSrc, uint32_t aLength) {
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NS_ABORT_IF_FALSE(mBuffer && mCapacity, "Buffer not initialized.");
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NS_ABORT_IF_FALSE(aLength <= Available(), "Buffer full.");
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uint32_t end = (mStart + mCount) % mCapacity;
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uint32_t toCopy = std::min(mCapacity - end, aLength);
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memcpy(&mBuffer[end], aSrc, toCopy);
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memcpy(&mBuffer[0], aSrc + toCopy, aLength - toCopy);
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mCount += aLength;
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}
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// Remove aSize bytes from the buffer. Caller must check returned size in
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// aSize{1,2} before using the pointer returned in aData{1,2}. Caller
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// must not specify an aSize larger than Length().
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void PopElements(uint32_t aSize, void** aData1, uint32_t* aSize1,
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void** aData2, uint32_t* aSize2) {
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NS_ABORT_IF_FALSE(mBuffer && mCapacity, "Buffer not initialized.");
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NS_ABORT_IF_FALSE(aSize <= Length(), "Request too large.");
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*aData1 = &mBuffer[mStart];
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*aSize1 = std::min(mCapacity - mStart, aSize);
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*aData2 = &mBuffer[0];
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*aSize2 = aSize - *aSize1;
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mCount -= *aSize1 + *aSize2;
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mStart += *aSize1 + *aSize2;
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mStart %= mCapacity;
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}
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// Throw away all but aSize bytes from the buffer. Returns new size, which
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// may be less than aSize
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uint32_t ContractTo(uint32_t aSize) {
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NS_ABORT_IF_FALSE(mBuffer && mCapacity, "Buffer not initialized.");
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if (aSize >= mCount) {
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return mCount;
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}
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mStart += (mCount - aSize);
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mCount = aSize;
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mStart %= mCapacity;
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return mCount;
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}
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size_t SizeOfExcludingThis(MallocSizeOf aMallocSizeOf) const
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{
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size_t amount = 0;
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amount += mBuffer.SizeOfExcludingThis(aMallocSizeOf);
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return amount;
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}
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private:
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nsAutoArrayPtr<uint8_t> mBuffer;
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uint32_t mCapacity;
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uint32_t mStart;
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uint32_t mCount;
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};
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class AudioInitTask;
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// Access to a single instance of this class must be synchronized by
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// callers, or made from a single thread. One exception is that access to
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// GetPosition, GetPositionInFrames, SetVolume, and Get{Rate,Channels}
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// is thread-safe without external synchronization.
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class AudioStream MOZ_FINAL
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{
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public:
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// Initialize Audio Library. Some Audio backends require initializing the
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// library before using it.
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static void InitLibrary();
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// Shutdown Audio Library. Some Audio backends require shutting down the
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// library after using it.
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static void ShutdownLibrary();
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// Returns the maximum number of channels supported by the audio hardware.
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static int MaxNumberOfChannels();
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// Queries the samplerate the hardware/mixer runs at, and stores it.
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// Can be called on any thread. When this returns, it is safe to call
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// PreferredSampleRate without locking.
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static void InitPreferredSampleRate();
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// Get the aformentionned sample rate. Does not lock.
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static int PreferredSampleRate();
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NS_INLINE_DECL_THREADSAFE_REFCOUNTING(AudioStream)
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AudioStream();
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virtual ~AudioStream();
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enum LatencyRequest {
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HighLatency,
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LowLatency
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};
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// Initialize the audio stream. aNumChannels is the number of audio
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// channels (1 for mono, 2 for stereo, etc) and aRate is the sample rate
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// (22050Hz, 44100Hz, etc).
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nsresult Init(int32_t aNumChannels, int32_t aRate,
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const dom::AudioChannel aAudioStreamChannel,
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LatencyRequest aLatencyRequest);
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// Closes the stream. All future use of the stream is an error.
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void Shutdown();
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// Write audio data to the audio hardware. aBuf is an array of AudioDataValues
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// AudioDataValue of length aFrames*mChannels. If aFrames is larger
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// than the result of Available(), the write will block until sufficient
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// buffer space is available. aTime is the time in ms associated with the first sample
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// for latency calculations
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nsresult Write(const AudioDataValue* aBuf, uint32_t aFrames, TimeStamp* aTime = nullptr);
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// Return the number of audio frames that can be written without blocking.
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uint32_t Available();
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// Set the current volume of the audio playback. This is a value from
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// 0 (meaning muted) to 1 (meaning full volume). Thread-safe.
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void SetVolume(double aVolume);
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// Block until buffered audio data has been consumed.
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void Drain();
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// Start the stream.
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void Start();
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// Return the number of frames written so far in the stream. This allow the
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// caller to check if it is safe to start the stream, if needed.
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int64_t GetWritten();
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// Pause audio playback.
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void Pause();
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// Resume audio playback.
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void Resume();
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// Return the position in microseconds of the audio frame being played by
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// the audio hardware, compensated for playback rate change. Thread-safe.
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int64_t GetPosition();
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// Return the position, measured in audio frames played since the stream
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// was opened, of the audio hardware. Thread-safe.
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int64_t GetPositionInFrames();
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// Return the position, measured in audio framed played since the stream was
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// opened, of the audio hardware, not adjusted for the changes of playback
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// rate.
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int64_t GetPositionInFramesInternal();
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// Returns true when the audio stream is paused.
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bool IsPaused();
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int GetRate() { return mOutRate; }
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int GetChannels() { return mChannels; }
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int GetOutChannels() { return mOutChannels; }
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// Set playback rate as a multiple of the intrinsic playback rate. This is to
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// be called only with aPlaybackRate > 0.0.
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nsresult SetPlaybackRate(double aPlaybackRate);
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// Switch between resampling (if false) and time stretching (if true, default).
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nsresult SetPreservesPitch(bool aPreservesPitch);
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size_t SizeOfIncludingThis(MallocSizeOf aMallocSizeOf) const;
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protected:
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friend class AudioClock;
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// Shared implementation of underflow adjusted position calculation.
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// Caller must own the monitor.
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int64_t GetPositionInFramesUnlocked();
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private:
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friend class AudioInitTask;
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// So we can call it asynchronously from AudioInitTask
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nsresult OpenCubeb(cubeb_stream_params &aParams,
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LatencyRequest aLatencyRequest);
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void CheckForStart();
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static void PrefChanged(const char* aPref, void* aClosure);
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static double GetVolumeScale();
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static bool GetFirstStream();
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static cubeb* GetCubebContext();
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static cubeb* GetCubebContextUnlocked();
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static uint32_t GetCubebLatency();
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static bool CubebLatencyPrefSet();
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static long DataCallback_S(cubeb_stream*, void* aThis, void* aBuffer, long aFrames)
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{
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return static_cast<AudioStream*>(aThis)->DataCallback(aBuffer, aFrames);
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}
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static void StateCallback_S(cubeb_stream*, void* aThis, cubeb_state aState)
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{
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static_cast<AudioStream*>(aThis)->StateCallback(aState);
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}
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long DataCallback(void* aBuffer, long aFrames);
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void StateCallback(cubeb_state aState);
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nsresult EnsureTimeStretcherInitializedUnlocked();
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// aTime is the time in ms the samples were inserted into MediaStreamGraph
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long GetUnprocessed(void* aBuffer, long aFrames, int64_t &aTime);
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long GetTimeStretched(void* aBuffer, long aFrames, int64_t &aTime);
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long GetUnprocessedWithSilencePadding(void* aBuffer, long aFrames, int64_t &aTime);
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int64_t GetLatencyInFrames();
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void GetBufferInsertTime(int64_t &aTimeMs);
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void StartUnlocked();
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// The monitor is held to protect all access to member variables. Write()
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// waits while mBuffer is full; DataCallback() notifies as it consumes
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// data from mBuffer. Drain() waits while mState is DRAINING;
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// StateCallback() notifies when mState is DRAINED.
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Monitor mMonitor;
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// Input rate in Hz (characteristic of the media being played)
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int mInRate;
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// Output rate in Hz (characteristic of the playback rate)
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int mOutRate;
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int mChannels;
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int mOutChannels;
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// Number of frames written to the buffers.
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int64_t mWritten;
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AudioClock mAudioClock;
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nsAutoPtr<soundtouch::SoundTouch> mTimeStretcher;
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nsRefPtr<AsyncLatencyLogger> mLatencyLog;
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// copy of Latency logger's starting time for offset calculations
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TimeStamp mStartTime;
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// Whether we are playing a low latency stream, or a normal stream.
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LatencyRequest mLatencyRequest;
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// Where in the current mInserts[0] block cubeb has read to
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int64_t mReadPoint;
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// Keep track of each inserted block of samples and the time it was inserted
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// so we can estimate the clock time for a specific sample's insertion (for when
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// we send data to cubeb). Blocks are aged out as needed.
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struct Inserts {
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int64_t mTimeMs;
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int64_t mFrames;
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};
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nsAutoTArray<Inserts, 8> mInserts;
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// Suppose we have received DataCallback for N times, |mWrittenFramesPast|
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// and |mLostFramesPast| are the sum of frames written to the backend from
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// 1st to |N-1|th DataCallbacks.
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uint64_t mWrittenFramesPast; // non-silent frames
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uint64_t mLostFramesPast; // silent frames
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// Frames written to the backend in Nth DataCallback.
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uint64_t mWrittenFramesLast; // non-silent frames
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uint64_t mLostFramesLast; // silent frames
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// Output file for dumping audio
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FILE* mDumpFile;
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// Temporary audio buffer. Filled by Write() and consumed by
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// DataCallback(). Once mBuffer is full, Write() blocks until sufficient
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// space becomes available in mBuffer. mBuffer is sized in bytes, not
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// frames.
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CircularByteBuffer mBuffer;
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// Software volume level. Applied during the servicing of DataCallback().
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double mVolume;
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// Owning reference to a cubeb_stream. cubeb_stream_destroy is called by
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// nsAutoRef's destructor.
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nsAutoRef<cubeb_stream> mCubebStream;
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uint32_t mBytesPerFrame;
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uint32_t BytesToFrames(uint32_t aBytes) {
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NS_ASSERTION(aBytes % mBytesPerFrame == 0,
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"Byte count not aligned on frames size.");
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return aBytes / mBytesPerFrame;
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}
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uint32_t FramesToBytes(uint32_t aFrames) {
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return aFrames * mBytesPerFrame;
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}
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enum StreamState {
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INITIALIZED, // Initialized, playback has not begun.
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STARTED, // cubeb started, but callbacks haven't started
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RUNNING, // DataCallbacks have started after STARTED, or after Resume().
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STOPPED, // Stopped by a call to Pause().
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DRAINING, // Drain requested. DataCallback will indicate end of stream
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// once the remaining contents of mBuffer are requested by
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// cubeb, after which StateCallback will indicate drain
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// completion.
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DRAINED, // StateCallback has indicated that the drain is complete.
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ERRORED, // Stream disabled due to an internal error.
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SHUTDOWN // Shutdown has been called
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};
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StreamState mState;
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bool mNeedsStart; // needed in case Start() is called before cubeb is open
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bool mIsFirst;
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// This mutex protects the static members below.
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static StaticMutex sMutex;
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static cubeb* sCubebContext;
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// Prefered samplerate, in Hz (characteristic of the
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// hardware/mixer/platform/API used).
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static uint32_t sPreferredSampleRate;
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static double sVolumeScale;
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static uint32_t sCubebLatency;
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static bool sCubebLatencyPrefSet;
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};
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class AudioInitTask : public nsRunnable
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{
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public:
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AudioInitTask(AudioStream *aStream,
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AudioStream::LatencyRequest aLatencyRequest,
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const cubeb_stream_params &aParams)
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: mAudioStream(aStream)
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, mLatencyRequest(aLatencyRequest)
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, mParams(aParams)
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{}
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nsresult Dispatch()
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{
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// Can't add 'this' as the event to run, since mThread may not be set yet
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nsresult rv = NS_NewNamedThread("CubebInit", getter_AddRefs(mThread));
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if (NS_SUCCEEDED(rv)) {
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// Note: event must not null out mThread!
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rv = mThread->Dispatch(this, NS_DISPATCH_NORMAL);
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}
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return rv;
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}
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protected:
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virtual ~AudioInitTask() {};
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private:
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NS_IMETHOD Run() MOZ_OVERRIDE MOZ_FINAL;
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RefPtr<AudioStream> mAudioStream;
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AudioStream::LatencyRequest mLatencyRequest;
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cubeb_stream_params mParams;
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nsCOMPtr<nsIThread> mThread;
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};
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} // namespace mozilla
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#endif
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