зеркало из https://github.com/mozilla/gecko-dev.git
387 строки
13 KiB
C++
387 строки
13 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 "CubebUtils.h"
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# include "MediaInfo.h"
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# include "MediaSink.h"
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# include "mozilla/Atomics.h"
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# include "mozilla/Monitor.h"
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# include "mozilla/MozPromise.h"
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# include "mozilla/ProfilerUtils.h"
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# include "mozilla/RefPtr.h"
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# include "mozilla/Result.h"
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# include "mozilla/TimeStamp.h"
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# include "mozilla/UniquePtr.h"
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# include "mozilla/SPSCQueue.h"
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# include "nsCOMPtr.h"
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# include "nsThreadUtils.h"
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# include "WavDumper.h"
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namespace soundtouch {
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class MOZ_EXPORT SoundTouch;
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}
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namespace mozilla {
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struct CubebDestroyPolicy {
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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 AudioConfig;
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// A struct that contains the number of frames serviced or underrun by a
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// callback, alongside the sample-rate for this callback (in case of playback
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// rate change, it can be variable).
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struct CallbackInfo {
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CallbackInfo() = default;
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CallbackInfo(uint32_t aServiced, uint32_t aUnderrun, uint32_t aOutputRate)
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: mServiced(aServiced), mUnderrun(aUnderrun), mOutputRate(aOutputRate) {}
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uint32_t mServiced = 0;
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uint32_t mUnderrun = 0;
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uint32_t mOutputRate = 0;
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};
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class AudioClock {
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public:
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explicit AudioClock(uint32_t aInRate);
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// Update the number of samples that has been written in the audio backend.
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// Called on the audio thread only.
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void UpdateFrameHistory(uint32_t aServiced, uint32_t aUnderrun,
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bool aAudioThreadChanged);
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/**
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* @param aFrames The playback position in frames of the audio engine.
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* @return The playback position in frames of the stream,
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* adjusted by playback rate changes and underrun frames.
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*/
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int64_t GetPositionInFrames(int64_t aFrames);
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/**
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* @param frames The playback position in frames of the audio engine.
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* @return The playback position in microseconds of the stream,
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* adjusted by playback rate changes and underrun frames.
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*/
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int64_t GetPosition(int64_t frames);
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// Set the playback rate.
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// Called on the audio thread only.
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void SetPlaybackRate(double aPlaybackRate);
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// Get the current playback rate.
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// Called on the audio thread only.
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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 only.
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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 only.
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bool GetPreservesPitch() const;
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// Called on either thread.
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uint32_t GetInputRate() const { return mInRate; }
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uint32_t GetOutputRate() const { return mOutRate; }
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private:
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// Output rate in Hz (characteristic of the playback rate). Written on the
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// audio thread, read on either thread.
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Atomic<uint32_t> mOutRate;
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// Input rate in Hz (characteristic of the media being played).
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const uint32_t mInRate;
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// True if the we are timestretching, false if we are resampling. Accessed on
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// the audio thread only.
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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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// Only accessed from non-audio threads on macOS, accessed on both threads and
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// protected by the AudioStream monitor on other platforms.
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const UniquePtr<FrameHistory> mFrameHistory
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# ifndef XP_MACOSX
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GUARDED_BY(mMutex)
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# endif
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;
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# ifdef XP_MACOSX
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// Enqueued on the audio thread, dequeued from the other thread. The maximum
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// size of this queue has been chosen empirically.
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SPSCQueue<CallbackInfo> mCallbackInfoQueue{100};
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// If it isn't possible to send the callback info to the non-audio thread,
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// store them here until it's possible to send them. This is an unlikely
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// fallback path. The size of this array has been chosen empirically. Only
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// ever accessed on the audio thread.
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AutoTArray<CallbackInfo, 5> mAudioThreadCallbackInfo;
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# else
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Mutex mMutex{"AudioClock"};
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# endif
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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(Span<AudioDataValue> aSpan, uint32_t aChannels,
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uint32_t aFrames)
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: mChannels(aChannels), mSpan(aSpan), 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_DIAGNOSTIC_ASSERT(Contains(aFrames));
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MOZ_ASSERT(mFrames >= aFrames);
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mFrames -= aFrames;
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mOffset += 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 {
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MOZ_DIAGNOSTIC_ASSERT(mOffset <= mSpan.Length());
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return mSpan.Elements() + mOffset;
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}
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protected:
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bool Contains(uint32_t aFrames) const {
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return mSpan.Length() >= mOffset + mChannels * aFrames;
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}
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const uint32_t mChannels;
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private:
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const Span<AudioDataValue> mSpan;
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size_t mOffset = 0;
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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 : public AudioBufferCursor {
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public:
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AudioBufferWriter(Span<AudioDataValue> aSpan, uint32_t aChannels,
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uint32_t aFrames)
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: AudioBufferCursor(aSpan, aChannels, aFrames) {}
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uint32_t WriteZeros(uint32_t aFrames) {
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MOZ_DIAGNOSTIC_ASSERT(Contains(aFrames));
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memset(Ptr(), 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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MOZ_DIAGNOSTIC_ASSERT(Contains(aFrames));
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memcpy(Ptr(), 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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MOZ_DIAGNOSTIC_ASSERT(Contains(aFrames));
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return Advance(aFunction(Ptr(), 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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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() = default;
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};
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class DataSource {
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public:
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// Attempt to acquire aFrames frames of audio, and returns the number of
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// frames successfuly acquired.
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virtual uint32_t PopFrames(AudioDataValue* aAudio, uint32_t aFrames,
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bool aAudioThreadChanged) = 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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protected:
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virtual ~DataSource() = default;
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};
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// aOutputChannels is the number of audio channels (1 for mono, 2 for stereo,
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// etc), aChannelMap is the indicator for channel layout(mono, stereo, 5.1 or
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// 7.1 ). Initialize the audio stream.and aRate is the sample rate
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// (22050Hz, 44100Hz, etc).
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AudioStream(DataSource& aSource, uint32_t aInRate, uint32_t aOutputChannels,
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AudioConfig::ChannelLayout::ChannelMap aChannelMap);
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nsresult Init(AudioDeviceInfo* aSinkInfo);
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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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void SetStreamName(const nsAString& aStreamName);
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// Start the stream and return a promise that will be resolve when the
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// playback completes.
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Result<already_AddRefed<MediaSink::EndedPromise>, nsresult> 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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static uint32_t GetPreferredRate() {
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return CubebUtils::PreferredSampleRate();
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}
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uint32_t GetOutChannels() const { return mOutChannels; }
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// Set playback rate as a multiple of the intrinsic playback rate. This is
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// to 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,
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// default).
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nsresult SetPreservesPitch(bool aPreservesPitch);
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size_t SizeOfIncludingThis(MallocSizeOf aMallocSizeOf) const;
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bool IsPlaybackCompleted() 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* aContext, cubeb_stream_params& aParams,
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TimeStamp aStartTime, bool aIsFirst);
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static long DataCallback_S(cubeb_stream*, void* aThis,
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const void* /* aInputBuffer */,
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void* aOutputBuffer, long aFrames) {
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return static_cast<AudioStream*>(aThis)->DataCallback(aOutputBuffer,
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aFrames);
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}
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static void StateCallback_S(cubeb_stream*, void* aThis, cubeb_state aState) {
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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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// Audio thread only
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nsresult EnsureTimeStretcherInitialized();
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void GetUnprocessed(AudioBufferWriter& aWriter);
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void GetTimeStretched(AudioBufferWriter& aWriter);
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void UpdatePlaybackRateIfNeeded();
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// Return true if audio frames are valid (correct sampling rate and valid
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// channel count) otherwise false.
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bool IsValidAudioFormat(Chunk* aChunk) REQUIRES(mMonitor);
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template <typename Function, typename... Args>
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int InvokeCubeb(Function aFunction, Args&&... aArgs) REQUIRES(mMonitor);
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bool CheckThreadIdChanged();
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void AssertIsOnAudioThread() const;
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soundtouch::SoundTouch* mTimeStretcher;
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AudioClock mAudioClock;
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WavDumper mDumpFile;
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const AudioConfig::ChannelLayout::ChannelMap mChannelMap;
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// The monitor is held to protect all access to member variables below.
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Monitor mMonitor MOZ_UNANNOTATED;
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const uint32_t mOutChannels;
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// Owning reference to a cubeb_stream. Set in Init(), cleared in Shutdown, so
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// no lock is needed to access.
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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.
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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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std::atomic<StreamState> mState;
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// DataSource::PopFrames can never be called concurrently.
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// DataSource::IsEnded uses only atomics.
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DataSource& mDataSource;
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// The device info of the current sink. If null
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// the default device is used. It is set
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// during the Init() in decoder thread.
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RefPtr<AudioDeviceInfo> mSinkInfo;
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// Contains the id of the audio thread, from profiler_get_thread_id.
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std::atomic<ProfilerThreadId> mAudioThreadId;
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const bool mSandboxed = false;
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MozPromiseHolder<MediaSink::EndedPromise> mEndedPromise GUARDED_BY(mMonitor);
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std::atomic<bool> mPlaybackComplete;
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// Both written on the MDSM thread, read on the audio thread.
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std::atomic<float> mPlaybackRate;
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std::atomic<bool> mPreservesPitch;
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// Audio thread only
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bool mAudioThreadChanged = false;
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};
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} // namespace mozilla
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#endif
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