зеркало из https://github.com/mozilla/pjs.git
747 строки
18 KiB
C++
747 строки
18 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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/* ***** BEGIN LICENSE BLOCK *****
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* Version: MPL 1.1/GPL 2.0/LGPL 2.1
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*
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* The contents of this file are subject to the Mozilla Public License Version
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* 1.1 (the "License"); you may not use this file except in compliance with
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* the License. You may obtain a copy of the License at
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* http://www.mozilla.org/MPL/
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*
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* Software distributed under the License is distributed on an "AS IS" basis,
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* WITHOUT WARRANTY OF ANY KIND, either express or implied. See the License
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* for the specific language governing rights and limitations under the
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* License.
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*
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* The Original Code is Mozilla code.
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*
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* The Initial Developer of the Original Code is
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* the Mozilla Foundation.
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* Portions created by the Initial Developer are Copyright (C) 2007
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* the Initial Developer. All Rights Reserved.
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*
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* Contributor(s):
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* Chris Double <chris.double@double.co.nz>
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*
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* Alternatively, the contents of this file may be used under the terms of
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* either the GNU General Public License Version 2 or later (the "GPL"), or
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* the GNU Lesser General Public License Version 2.1 or later (the "LGPL"),
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* in which case the provisions of the GPL or the LGPL are applicable instead
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* of those above. If you wish to allow use of your version of this file only
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* under the terms of either the GPL or the LGPL, and not to allow others to
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* use your version of this file under the terms of the MPL, indicate your
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* decision by deleting the provisions above and replace them with the notice
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* and other provisions required by the GPL or the LGPL. If you do not delete
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* the provisions above, a recipient may use your version of this file under
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* the terms of any one of the MPL, the GPL or the LGPL.
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*
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* ***** END LICENSE BLOCK ***** */
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#include "mozilla/dom/ContentChild.h"
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#include "mozilla/dom/PAudioChild.h"
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#include "mozilla/dom/AudioChild.h"
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#include "nsXULAppAPI.h"
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using namespace mozilla::dom;
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#include <stdio.h>
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#include <math.h>
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#include "prlog.h"
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#include "prmem.h"
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#include "nsAutoPtr.h"
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#include "nsAudioStream.h"
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#include "nsAlgorithm.h"
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#include "VideoUtils.h"
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extern "C" {
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#include "sydneyaudio/sydney_audio.h"
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}
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#include "mozilla/TimeStamp.h"
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#include "nsThreadUtils.h"
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#if defined(XP_MACOSX)
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#define SA_PER_STREAM_VOLUME 1
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#endif
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// Android's audio backend is not available in content processes, so audio must
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// be remoted to the parent chrome process.
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#if defined(ANDROID)
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#define REMOTE_AUDIO 1
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#endif
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using mozilla::TimeStamp;
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#ifdef PR_LOGGING
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PRLogModuleInfo* gAudioStreamLog = nsnull;
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#endif
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#define FAKE_BUFFER_SIZE 176400
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class nsAudioStreamLocal : public nsAudioStream
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{
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public:
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NS_DECL_ISUPPORTS
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~nsAudioStreamLocal();
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nsAudioStreamLocal();
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nsresult Init(PRInt32 aNumChannels, PRInt32 aRate, SampleFormat aFormat);
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void Shutdown();
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nsresult Write(const void* aBuf, PRUint32 aCount, PRBool aBlocking);
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PRUint32 Available();
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void SetVolume(double aVolume);
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void Drain();
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void Pause();
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void Resume();
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PRInt64 GetPosition();
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PRInt64 GetSampleOffset();
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PRBool IsPaused();
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PRInt32 GetMinWriteSamples();
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private:
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double mVolume;
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void* mAudioHandle;
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int mRate;
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int mChannels;
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SampleFormat mFormat;
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// When a Write() request is made, and the number of samples
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// requested to be written exceeds the buffer size of the audio
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// backend, the remaining samples are stored in this variable. They
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// will be written on the next Write() request.
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nsTArray<short> mBufferOverflow;
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// PR_TRUE if this audio stream is paused.
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PRPackedBool mPaused;
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// PR_TRUE if this stream has encountered an error.
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PRPackedBool mInError;
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};
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class nsAudioStreamRemote : public nsAudioStream
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{
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public:
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NS_DECL_ISUPPORTS
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nsAudioStreamRemote();
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~nsAudioStreamRemote();
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nsresult Init(PRInt32 aNumChannels, PRInt32 aRate, SampleFormat aFormat);
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void Shutdown();
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nsresult Write(const void* aBuf, PRUint32 aCount, PRBool aBlocking);
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PRUint32 Available();
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void SetVolume(double aVolume);
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void Drain();
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void Pause();
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void Resume();
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PRInt64 GetPosition();
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PRInt64 GetSampleOffset();
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PRBool IsPaused();
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PRInt32 GetMinWriteSamples();
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private:
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nsRefPtr<AudioChild> mAudioChild;
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SampleFormat mFormat;
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int mRate;
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int mChannels;
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PRInt32 mBytesPerSample;
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// PR_TRUE if this audio stream is paused.
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PRPackedBool mPaused;
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friend class AudioInitEvent;
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};
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class AudioInitEvent : public nsRunnable
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{
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public:
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AudioInitEvent(nsAudioStreamRemote* owner)
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{
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mOwner = owner;
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}
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NS_IMETHOD Run()
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{
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ContentChild * cpc = ContentChild::GetSingleton();
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NS_ASSERTION(cpc, "Content Protocol is NULL!");
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mOwner->mAudioChild = static_cast<AudioChild*> (cpc->SendPAudioConstructor(mOwner->mChannels,
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mOwner->mRate,
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mOwner->mFormat));
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return NS_OK;
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}
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nsRefPtr<nsAudioStreamRemote> mOwner;
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};
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class AudioWriteEvent : public nsRunnable
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{
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public:
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AudioWriteEvent(AudioChild* aChild,
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const void* aBuf,
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PRUint32 aNumberOfSamples,
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PRUint32 aBytesPerSample)
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{
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mAudioChild = aChild;
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mBytesPerSample = aBytesPerSample;
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mBuffer.Assign((const char*)aBuf, aNumberOfSamples*aBytesPerSample);
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}
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NS_IMETHOD Run()
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{
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if (!mAudioChild->IsIPCOpen())
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return NS_OK;
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mAudioChild->SendWrite(mBuffer,
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mBuffer.Length() / mBytesPerSample);
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return NS_OK;
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}
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nsRefPtr<AudioChild> mAudioChild;
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nsCString mBuffer;
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PRUint32 mBytesPerSample;
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};
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class AudioSetVolumeEvent : public nsRunnable
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{
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public:
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AudioSetVolumeEvent(AudioChild* aChild, double aVolume)
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{
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mAudioChild = aChild;
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mVolume = aVolume;
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}
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NS_IMETHOD Run()
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{
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if (!mAudioChild->IsIPCOpen())
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return NS_OK;
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mAudioChild->SendSetVolume(mVolume);
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return NS_OK;
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}
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nsRefPtr<AudioChild> mAudioChild;
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double mVolume;
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};
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class AudioDrainEvent : public nsRunnable
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{
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public:
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AudioDrainEvent(AudioChild* aChild)
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{
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mAudioChild = aChild;
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}
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NS_IMETHOD Run()
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{
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if (!mAudioChild->IsIPCOpen())
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return NS_OK;
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mAudioChild->SendDrain();
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return NS_OK;
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}
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nsRefPtr<AudioChild> mAudioChild;
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};
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class AudioPauseEvent : public nsRunnable
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{
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public:
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AudioPauseEvent(AudioChild* aChild, PRBool pause)
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{
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mAudioChild = aChild;
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mPause = pause;
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}
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NS_IMETHOD Run()
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{
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if (!mAudioChild->IsIPCOpen())
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return NS_OK;
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if (mPause)
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mAudioChild->SendPause();
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else
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mAudioChild->SendResume();
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return NS_OK;
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}
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nsRefPtr<AudioChild> mAudioChild;
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PRBool mPause;
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};
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class AudioShutdownEvent : public nsRunnable
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{
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public:
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AudioShutdownEvent(AudioChild* aChild)
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{
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mAudioChild = aChild;
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}
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NS_IMETHOD Run()
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{
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if (mAudioChild->IsIPCOpen())
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mAudioChild->SendShutdown();
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return NS_OK;
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}
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nsRefPtr<AudioChild> mAudioChild;
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};
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void nsAudioStream::InitLibrary()
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{
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#ifdef PR_LOGGING
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gAudioStreamLog = PR_NewLogModule("nsAudioStream");
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#endif
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}
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void nsAudioStream::ShutdownLibrary()
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{
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}
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nsIThread *
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nsAudioStream::GetThread()
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{
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if (!mAudioPlaybackThread) {
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NS_NewThread(getter_AddRefs(mAudioPlaybackThread));
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}
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return mAudioPlaybackThread;
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}
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nsAudioStream* nsAudioStream::AllocateStream()
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{
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#if defined(REMOTE_AUDIO)
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if (XRE_GetProcessType() == GeckoProcessType_Content) {
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return new nsAudioStreamRemote();
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}
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#endif
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return new nsAudioStreamLocal();
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}
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class AsyncShutdownPlaybackThread : public nsRunnable
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{
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public:
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AsyncShutdownPlaybackThread(nsIThread* aThread) : mThread(aThread) {}
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NS_IMETHODIMP Run() { return mThread->Shutdown(); }
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private:
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nsCOMPtr<nsIThread> mThread;
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};
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nsAudioStream::~nsAudioStream()
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{
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if (mAudioPlaybackThread) {
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nsCOMPtr<nsIRunnable> event = new AsyncShutdownPlaybackThread(mAudioPlaybackThread);
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NS_DispatchToMainThread(event);
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}
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}
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nsAudioStreamLocal::nsAudioStreamLocal() :
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mVolume(1.0),
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mAudioHandle(0),
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mRate(0),
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mChannels(0),
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mFormat(FORMAT_S16_LE),
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mPaused(PR_FALSE),
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mInError(PR_FALSE)
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{
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}
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nsAudioStreamLocal::~nsAudioStreamLocal()
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{
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Shutdown();
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}
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NS_IMPL_THREADSAFE_ISUPPORTS0(nsAudioStreamLocal)
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nsresult nsAudioStreamLocal::Init(PRInt32 aNumChannels, PRInt32 aRate, SampleFormat aFormat)
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{
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mRate = aRate;
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mChannels = aNumChannels;
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mFormat = aFormat;
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if (sa_stream_create_pcm(reinterpret_cast<sa_stream_t**>(&mAudioHandle),
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NULL,
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SA_MODE_WRONLY,
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SA_PCM_FORMAT_S16_NE,
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aRate,
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aNumChannels) != SA_SUCCESS) {
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mAudioHandle = nsnull;
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mInError = PR_TRUE;
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PR_LOG(gAudioStreamLog, PR_LOG_ERROR, ("nsAudioStreamLocal: sa_stream_create_pcm error"));
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return NS_ERROR_FAILURE;
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}
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if (sa_stream_open(static_cast<sa_stream_t*>(mAudioHandle)) != SA_SUCCESS) {
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sa_stream_destroy(static_cast<sa_stream_t*>(mAudioHandle));
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mAudioHandle = nsnull;
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mInError = PR_TRUE;
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PR_LOG(gAudioStreamLog, PR_LOG_ERROR, ("nsAudioStreamLocal: sa_stream_open error"));
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return NS_ERROR_FAILURE;
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}
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mInError = PR_FALSE;
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return NS_OK;
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}
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void nsAudioStreamLocal::Shutdown()
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{
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if (!mAudioHandle)
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return;
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sa_stream_destroy(static_cast<sa_stream_t*>(mAudioHandle));
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mAudioHandle = nsnull;
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mInError = PR_TRUE;
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}
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nsresult nsAudioStreamLocal::Write(const void* aBuf, PRUint32 aCount, PRBool aBlocking)
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{
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NS_ABORT_IF_FALSE(aCount % mChannels == 0,
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"Buffer size must be divisible by channel count");
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NS_ASSERTION(!mPaused, "Don't write audio when paused, you'll block");
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if (mInError)
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return NS_ERROR_FAILURE;
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PRUint32 offset = mBufferOverflow.Length();
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PRUint32 count = aCount + offset;
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nsAutoArrayPtr<short> s_data(new short[count]);
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if (s_data) {
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for (PRUint32 i=0; i < offset; ++i) {
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s_data[i] = mBufferOverflow.ElementAt(i);
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}
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mBufferOverflow.Clear();
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switch (mFormat) {
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case FORMAT_U8: {
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const PRUint8* buf = static_cast<const PRUint8*>(aBuf);
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PRInt32 volume = PRInt32((1 << 16) * mVolume);
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for (PRUint32 i = 0; i < aCount; ++i) {
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s_data[i + offset] = short(((PRInt32(buf[i]) - 128) * volume) >> 8);
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}
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break;
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}
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case FORMAT_S16_LE: {
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const short* buf = static_cast<const short*>(aBuf);
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PRInt32 volume = PRInt32((1 << 16) * mVolume);
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for (PRUint32 i = 0; i < aCount; ++i) {
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short s = buf[i];
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#if defined(IS_BIG_ENDIAN)
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s = ((s & 0x00ff) << 8) | ((s & 0xff00) >> 8);
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#endif
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s_data[i + offset] = short((PRInt32(s) * volume) >> 16);
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}
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break;
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}
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case FORMAT_FLOAT32: {
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const float* buf = static_cast<const float*>(aBuf);
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for (PRUint32 i = 0; i < aCount; ++i) {
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float scaled_value = floorf(0.5 + 32768 * buf[i] * mVolume);
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if (buf[i] < 0.0) {
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s_data[i + offset] = (scaled_value < -32768.0) ?
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-32768 :
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short(scaled_value);
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} else {
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s_data[i+offset] = (scaled_value > 32767.0) ?
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32767 :
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short(scaled_value);
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}
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}
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break;
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}
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}
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if (!aBlocking) {
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// We're running in non-blocking mode, crop the data to the amount
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// which is available in the audio buffer, and save the rest for
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// subsequent calls.
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PRUint32 available = Available();
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if (available < count) {
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mBufferOverflow.AppendElements(s_data.get() + available, (count - available));
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count = available;
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}
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}
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if (sa_stream_write(static_cast<sa_stream_t*>(mAudioHandle),
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s_data.get(),
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count * sizeof(short)) != SA_SUCCESS)
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{
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PR_LOG(gAudioStreamLog, PR_LOG_ERROR, ("nsAudioStreamLocal: sa_stream_write error"));
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mInError = PR_TRUE;
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return NS_ERROR_FAILURE;
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}
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}
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return NS_OK;
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}
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PRUint32 nsAudioStreamLocal::Available()
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{
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// If the audio backend failed to open, lie and say we'll accept some
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// data.
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if (mInError)
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return FAKE_BUFFER_SIZE;
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size_t s = 0;
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if (sa_stream_get_write_size(static_cast<sa_stream_t*>(mAudioHandle), &s) != SA_SUCCESS)
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return 0;
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return s / sizeof(short);
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}
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void nsAudioStreamLocal::SetVolume(double aVolume)
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{
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NS_ASSERTION(aVolume >= 0.0 && aVolume <= 1.0, "Invalid volume");
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#if defined(SA_PER_STREAM_VOLUME)
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if (sa_stream_set_volume_abs(static_cast<sa_stream_t*>(mAudioHandle), aVolume) != SA_SUCCESS) {
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PR_LOG(gAudioStreamLog, PR_LOG_ERROR, ("nsAudioStreamLocal: sa_stream_set_volume_abs error"));
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mInError = PR_TRUE;
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}
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#else
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mVolume = aVolume;
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#endif
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}
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void nsAudioStreamLocal::Drain()
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{
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NS_ASSERTION(!mPaused, "Don't drain audio when paused, it won't finish!");
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if (mInError)
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return;
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// Write any remaining unwritten sound data in the overflow buffer
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if (!mBufferOverflow.IsEmpty()) {
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if (sa_stream_write(static_cast<sa_stream_t*>(mAudioHandle),
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mBufferOverflow.Elements(),
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mBufferOverflow.Length() * sizeof(short)) != SA_SUCCESS)
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PR_LOG(gAudioStreamLog, PR_LOG_ERROR, ("nsAudioStreamLocal: sa_stream_write error"));
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mInError = PR_TRUE;
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return;
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}
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int r = sa_stream_drain(static_cast<sa_stream_t*>(mAudioHandle));
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if (r != SA_SUCCESS && r != SA_ERROR_INVALID) {
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PR_LOG(gAudioStreamLog, PR_LOG_ERROR, ("nsAudioStreamLocal: sa_stream_drain error"));
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mInError = PR_TRUE;
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}
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}
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void nsAudioStreamLocal::Pause()
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{
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if (mInError)
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return;
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mPaused = PR_TRUE;
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sa_stream_pause(static_cast<sa_stream_t*>(mAudioHandle));
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}
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void nsAudioStreamLocal::Resume()
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{
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if (mInError)
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return;
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mPaused = PR_FALSE;
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sa_stream_resume(static_cast<sa_stream_t*>(mAudioHandle));
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}
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PRInt64 nsAudioStreamLocal::GetPosition()
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{
|
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PRInt64 sampleOffset = GetSampleOffset();
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if (sampleOffset >= 0) {
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return ((USECS_PER_S * sampleOffset) / mRate / mChannels);
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}
|
|
return -1;
|
|
}
|
|
|
|
PRInt64 nsAudioStreamLocal::GetSampleOffset()
|
|
{
|
|
if (mInError) {
|
|
return -1;
|
|
}
|
|
|
|
sa_position_t positionType = SA_POSITION_WRITE_SOFTWARE;
|
|
#if defined(XP_WIN)
|
|
positionType = SA_POSITION_WRITE_HARDWARE;
|
|
#endif
|
|
int64_t position = 0;
|
|
if (sa_stream_get_position(static_cast<sa_stream_t*>(mAudioHandle),
|
|
positionType, &position) == SA_SUCCESS) {
|
|
return position / sizeof(short);
|
|
}
|
|
|
|
return -1;
|
|
}
|
|
|
|
PRBool nsAudioStreamLocal::IsPaused()
|
|
{
|
|
return mPaused;
|
|
}
|
|
|
|
PRInt32 nsAudioStreamLocal::GetMinWriteSamples()
|
|
{
|
|
size_t samples;
|
|
int r = sa_stream_get_min_write(static_cast<sa_stream_t*>(mAudioHandle),
|
|
&samples);
|
|
if (r == SA_ERROR_NOT_SUPPORTED)
|
|
return 1;
|
|
else if (r != SA_SUCCESS || samples > PR_INT32_MAX)
|
|
return -1;
|
|
|
|
return static_cast<PRInt32>(samples);
|
|
}
|
|
|
|
nsAudioStreamRemote::nsAudioStreamRemote()
|
|
: mAudioChild(NULL),
|
|
mFormat(FORMAT_S16_LE),
|
|
mRate(0),
|
|
mChannels(0),
|
|
mBytesPerSample(1),
|
|
mPaused(PR_FALSE)
|
|
{}
|
|
|
|
nsAudioStreamRemote::~nsAudioStreamRemote()
|
|
{
|
|
Shutdown();
|
|
}
|
|
|
|
NS_IMPL_THREADSAFE_ISUPPORTS0(nsAudioStreamRemote)
|
|
|
|
nsresult
|
|
nsAudioStreamRemote::Init(PRInt32 aNumChannels,
|
|
PRInt32 aRate,
|
|
SampleFormat aFormat)
|
|
{
|
|
mRate = aRate;
|
|
mChannels = aNumChannels;
|
|
mFormat = aFormat;
|
|
|
|
switch (mFormat) {
|
|
case FORMAT_U8: {
|
|
mBytesPerSample = sizeof(PRUint8);
|
|
break;
|
|
}
|
|
case FORMAT_S16_LE: {
|
|
mBytesPerSample = sizeof(short);
|
|
break;
|
|
}
|
|
case FORMAT_FLOAT32: {
|
|
mBytesPerSample = sizeof(float);
|
|
}
|
|
}
|
|
|
|
nsCOMPtr<nsIRunnable> event = new AudioInitEvent(this);
|
|
NS_DispatchToMainThread(event, NS_DISPATCH_SYNC);
|
|
return NS_OK;
|
|
}
|
|
|
|
void
|
|
nsAudioStreamRemote::Shutdown()
|
|
{
|
|
if (!mAudioChild)
|
|
return;
|
|
nsCOMPtr<nsIRunnable> event = new AudioShutdownEvent(mAudioChild);
|
|
NS_DispatchToMainThread(event);
|
|
mAudioChild = nsnull;
|
|
}
|
|
|
|
nsresult
|
|
nsAudioStreamRemote::Write(const void* aBuf,
|
|
PRUint32 aCount,
|
|
PRBool aBlocking)
|
|
{
|
|
if (!mAudioChild)
|
|
return NS_ERROR_FAILURE;
|
|
nsCOMPtr<nsIRunnable> event = new AudioWriteEvent(mAudioChild,
|
|
aBuf,
|
|
aCount,
|
|
mBytesPerSample);
|
|
NS_DispatchToMainThread(event);
|
|
return NS_OK;
|
|
}
|
|
|
|
PRUint32
|
|
nsAudioStreamRemote::Available()
|
|
{
|
|
return FAKE_BUFFER_SIZE;
|
|
}
|
|
|
|
PRInt32 nsAudioStreamRemote::GetMinWriteSamples()
|
|
{
|
|
/** TODO: Implement this function for remoting. We could potentially remote
|
|
to a backend which has a start threshold... */
|
|
return 1;
|
|
}
|
|
|
|
void
|
|
nsAudioStreamRemote::SetVolume(double aVolume)
|
|
{
|
|
if (!mAudioChild)
|
|
return;
|
|
nsCOMPtr<nsIRunnable> event = new AudioSetVolumeEvent(mAudioChild, aVolume);
|
|
NS_DispatchToMainThread(event);
|
|
}
|
|
|
|
void
|
|
nsAudioStreamRemote::Drain()
|
|
{
|
|
if (!mAudioChild)
|
|
return;
|
|
nsCOMPtr<nsIRunnable> event = new AudioDrainEvent(mAudioChild);
|
|
NS_DispatchToMainThread(event);
|
|
mAudioChild->WaitForDrain();
|
|
}
|
|
|
|
void
|
|
nsAudioStreamRemote::Pause()
|
|
{
|
|
mPaused = PR_TRUE;
|
|
if (!mAudioChild)
|
|
return;
|
|
nsCOMPtr<nsIRunnable> event = new AudioPauseEvent(mAudioChild, PR_TRUE);
|
|
NS_DispatchToMainThread(event);
|
|
}
|
|
|
|
void
|
|
nsAudioStreamRemote::Resume()
|
|
{
|
|
mPaused = PR_FALSE;
|
|
if (!mAudioChild)
|
|
return;
|
|
nsCOMPtr<nsIRunnable> event = new AudioPauseEvent(mAudioChild, PR_FALSE);
|
|
NS_DispatchToMainThread(event);
|
|
}
|
|
|
|
PRInt64 nsAudioStreamRemote::GetPosition()
|
|
{
|
|
PRInt64 sampleOffset = GetSampleOffset();
|
|
if (sampleOffset >= 0) {
|
|
return ((USECS_PER_S * sampleOffset) / mRate / mChannels);
|
|
}
|
|
return 0;
|
|
}
|
|
|
|
PRInt64
|
|
nsAudioStreamRemote::GetSampleOffset()
|
|
{
|
|
if(!mAudioChild)
|
|
return 0;
|
|
|
|
PRInt64 offset = mAudioChild->GetLastKnownSampleOffset();
|
|
if (offset == -1)
|
|
return 0;
|
|
|
|
PRInt64 time = mAudioChild->GetLastKnownSampleOffsetTime();
|
|
PRInt64 result = offset + (mRate * mChannels * (PR_IntervalNow() - time) / USECS_PER_S);
|
|
|
|
return result;
|
|
}
|
|
|
|
PRBool
|
|
nsAudioStreamRemote::IsPaused()
|
|
{
|
|
return mPaused;
|
|
}
|