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a853e094df
--HG-- extra : commitid : C4GE8epQXOe
382 lines
12 KiB
C++
382 lines
12 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 "nsCOMPtr.h"
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#include "nsThreadUtils.h"
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#include "mozilla/dom/AudioChannelBinding.h"
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#include "mozilla/Monitor.h"
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#include "mozilla/RefPtr.h"
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#include "mozilla/TimeStamp.h"
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#include "mozilla/UniquePtr.h"
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#include "CubebUtils.h"
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#include "soundtouch/SoundTouchFactory.h"
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namespace mozilla {
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struct CubebDestroyPolicy
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{
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void operator()(cubeb_stream* aStream) const {
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cubeb_stream_destroy(aStream);
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}
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};
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class AudioStream;
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class FrameHistory;
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class AudioClock
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{
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public:
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explicit 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 UpdateFrameHistory(uint32_t aServiced, uint32_t aUnderrun);
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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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int64_t GetPositionUnlocked() const;
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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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int64_t GetPositionInFrames() const;
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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() const;
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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() const;
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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* const mAudioStream;
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// Output rate in Hz (characteristic of the playback rate)
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uint32_t mOutRate;
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// Input rate in Hz (characteristic of the media being played)
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uint32_t 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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// The history of frames sent to the audio engine in each DataCallback.
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const nsAutoPtr<FrameHistory> mFrameHistory;
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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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MOZ_ASSERT(!mBuffer, "Buffer allocated.");
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mCapacity = aCapacity;
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mBuffer = MakeUnique<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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MOZ_ASSERT(mBuffer && mCapacity, "Buffer not initialized.");
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MOZ_ASSERT(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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MOZ_ASSERT(mBuffer && mCapacity, "Buffer not initialized.");
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MOZ_ASSERT(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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size_t SizeOfExcludingThis(MallocSizeOf aMallocSizeOf) const
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{
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size_t amount = 0;
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amount += aMallocSizeOf(mBuffer.get());
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return amount;
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}
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private:
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UniquePtr<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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/*
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* A bookkeeping class to track the read/write position of an audio buffer.
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*/
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class AudioBufferCursor {
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public:
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AudioBufferCursor(AudioDataValue* aPtr, uint32_t aChannels, uint32_t aFrames)
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: mPtr(aPtr), mChannels(aChannels), mFrames(aFrames) {}
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// Advance the cursor to account for frames that are consumed.
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uint32_t Advance(uint32_t aFrames) {
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MOZ_ASSERT(mFrames >= aFrames);
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mFrames -= aFrames;
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mPtr += mChannels * aFrames;
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return aFrames;
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}
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// The number of frames available for read/write in this buffer.
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uint32_t Available() const { return mFrames; }
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// Return a pointer where read/write should begin.
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AudioDataValue* Ptr() const { return mPtr; }
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protected:
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AudioDataValue* mPtr;
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const uint32_t mChannels;
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uint32_t mFrames;
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};
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/*
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* A helper class to encapsulate pointer arithmetic and provide means to modify
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* the underlying audio buffer.
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*/
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class AudioBufferWriter : private AudioBufferCursor {
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public:
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AudioBufferWriter(AudioDataValue* aPtr, uint32_t aChannels, uint32_t aFrames)
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: AudioBufferCursor(aPtr, aChannels, aFrames) {}
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uint32_t WriteZeros(uint32_t aFrames) {
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memset(mPtr, 0, sizeof(AudioDataValue) * mChannels * aFrames);
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return Advance(aFrames);
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}
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uint32_t Write(const AudioDataValue* aPtr, uint32_t aFrames) {
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memcpy(mPtr, aPtr, sizeof(AudioDataValue) * mChannels * aFrames);
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return Advance(aFrames);
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}
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// Provide a write fuction to update the audio buffer with the following
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// signature: uint32_t(const AudioDataValue* aPtr, uint32_t aFrames)
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// aPtr: Pointer to the audio buffer.
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// aFrames: The number of frames available in the buffer.
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// return: The number of frames actually written by the function.
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template <typename Function>
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uint32_t Write(const Function& aFunction, uint32_t aFrames) {
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return Advance(aFunction(mPtr, aFrames));
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}
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using AudioBufferCursor::Available;
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};
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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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// SetMicrophoneActive is thread-safe without external synchronization.
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class AudioStream final
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{
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virtual ~AudioStream();
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public:
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NS_INLINE_DECL_THREADSAFE_REFCOUNTING(AudioStream)
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class Chunk {
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public:
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// Return a pointer to the audio data.
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virtual const AudioDataValue* Data() const = 0;
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// Return the number of frames in this chunk.
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virtual uint32_t Frames() const = 0;
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// Return the number of audio channels.
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virtual uint32_t Channels() const = 0;
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// Return the sample rate of this chunk.
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virtual uint32_t Rate() const = 0;
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// Return a writable pointer for downmixing.
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virtual AudioDataValue* GetWritable() const = 0;
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virtual ~Chunk() {}
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};
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class DataSource {
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public:
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// Return a chunk which contains at most aFrames frames or zero if no
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// frames in the source at all.
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virtual UniquePtr<Chunk> PopFrames(uint32_t aFrames) = 0;
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// Return true if no more data will be added to the source.
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virtual bool Ended() const = 0;
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// Notify that all data is drained by the AudioStream.
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virtual void Drained() = 0;
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protected:
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virtual ~DataSource() {}
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};
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explicit AudioStream(DataSource& aSource);
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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(uint32_t aNumChannels, uint32_t aRate,
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const dom::AudioChannel aAudioStreamChannel);
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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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void Reset();
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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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// Start the stream.
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void Start();
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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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// Returns true when the audio stream is paused.
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bool IsPaused();
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uint32_t GetRate() { return mOutRate; }
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uint32_t GetChannels() { return mChannels; }
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uint32_t 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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// Return the position, measured in audio frames 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 or underrun frames.
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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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nsresult OpenCubeb(cubeb_stream_params &aParams);
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static long DataCallback_S(cubeb_stream*, void* aThis,
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const void* /* aInputBuffer */, void* aOutputBuffer,
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long aFrames)
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{
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return static_cast<AudioStream*>(aThis)->DataCallback(aOutputBuffer, 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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// Return true if downmixing succeeds otherwise false.
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bool Downmix(Chunk* aChunk);
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void GetUnprocessed(AudioBufferWriter& aWriter);
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void GetTimeStretched(AudioBufferWriter& aWriter);
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void StartUnlocked();
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// The monitor is held to protect all access to member variables.
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Monitor mMonitor;
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// Input rate in Hz (characteristic of the media being played)
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uint32_t mInRate;
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// Output rate in Hz (characteristic of the playback rate)
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uint32_t mOutRate;
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uint32_t mChannels;
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uint32_t mOutChannels;
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#if defined(__ANDROID__)
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dom::AudioChannel mAudioChannel;
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#endif
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AudioClock mAudioClock;
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soundtouch::SoundTouch* mTimeStretcher;
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// Stream start time for stream open delay telemetry.
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TimeStamp mStartTime;
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// Output file for dumping audio
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FILE* mDumpFile;
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// Owning reference to a cubeb_stream.
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UniquePtr<cubeb_stream, CubebDestroyPolicy> mCubebStream;
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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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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 mIsFirst;
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// Get this value from the preferece, if true, we would downmix the stereo.
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bool mIsMonoAudioEnabled;
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DataSource& mDataSource;
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};
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} // namespace mozilla
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#endif
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