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-rwxr-xr-xindra/llaudio/llaudiodecodemgr.cpp1602
-rw-r--r--indra/llaudio/llaudiodecodemgr.h108
-rwxr-xr-xindra/llaudio/llaudioengine.h972
-rw-r--r--indra/llaudio/llaudioengine_openal.cpp1120
-rw-r--r--indra/llaudio/llaudioengine_openal.h216
-rw-r--r--indra/llaudio/lllistener_openal.h118
-rw-r--r--indra/llaudio/llvorbisencode.cpp1012
7 files changed, 2574 insertions, 2574 deletions
diff --git a/indra/llaudio/llaudiodecodemgr.cpp b/indra/llaudio/llaudiodecodemgr.cpp
index 602d360f1c..b086e49ba4 100755
--- a/indra/llaudio/llaudiodecodemgr.cpp
+++ b/indra/llaudio/llaudiodecodemgr.cpp
@@ -1,801 +1,801 @@
-/**
- * @file llaudiodecodemgr.cpp
- *
- * $LicenseInfo:firstyear=2003&license=viewerlgpl$
- * Second Life Viewer Source Code
- * Copyright (C) 2010, Linden Research, Inc.
- *
- * This library is free software; you can redistribute it and/or
- * modify it under the terms of the GNU Lesser General Public
- * License as published by the Free Software Foundation;
- * version 2.1 of the License only.
- *
- * This library is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
- * Lesser General Public License for more details.
- *
- * You should have received a copy of the GNU Lesser General Public
- * License along with this library; if not, write to the Free Software
- * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
- *
- * Linden Research, Inc., 945 Battery Street, San Francisco, CA 94111 USA
- * $/LicenseInfo$
- */
-
-#include "linden_common.h"
-
-#include "llaudiodecodemgr.h"
-
-#include "llaudioengine.h"
-#include "lllfsthread.h"
-#include "llfilesystem.h"
-#include "llstring.h"
-#include "lldir.h"
-#include "llendianswizzle.h"
-#include "llassetstorage.h"
-#include "llrefcount.h"
-#include "threadpool.h"
-#include "workqueue.h"
-
-#include "llvorbisencode.h"
-
-#include "vorbis/codec.h"
-#include "vorbis/vorbisfile.h"
-#include <iterator>
-#include <deque>
-
-extern LLAudioEngine *gAudiop;
-
-static const S32 WAV_HEADER_SIZE = 44;
-
-
-//////////////////////////////////////////////////////////////////////////////
-
-
-class LLVorbisDecodeState : public LLThreadSafeRefCount
-{
-public:
- class WriteResponder : public LLLFSThread::Responder
- {
- public:
- WriteResponder(LLVorbisDecodeState* decoder) : mDecoder(decoder) {}
- ~WriteResponder() {}
- void completed(S32 bytes)
- {
- mDecoder->ioComplete(bytes);
- }
- LLPointer<LLVorbisDecodeState> mDecoder;
- };
-
- LLVorbisDecodeState(const LLUUID &uuid, const std::string &out_filename);
-
- bool initDecode();
- bool decodeSection(); // Return true if done.
- bool finishDecode();
-
- void flushBadFile();
-
- void ioComplete(S32 bytes) { mBytesRead = bytes; }
- bool isValid() const { return mValid; }
- bool isDone() const { return mDone; }
- const LLUUID &getUUID() const { return mUUID; }
-
-protected:
- virtual ~LLVorbisDecodeState();
-
- bool mValid;
- bool mDone;
- LLAtomicS32 mBytesRead;
- LLUUID mUUID;
-
- std::vector<U8> mWAVBuffer;
- std::string mOutFilename;
- LLLFSThread::handle_t mFileHandle;
-
- LLFileSystem *mInFilep;
- OggVorbis_File mVF;
- S32 mCurrentSection;
-};
-
-size_t cache_read(void *ptr, size_t size, size_t nmemb, void *datasource)
-{
- LLFileSystem *file = (LLFileSystem *)datasource;
-
- if (file->read((U8*)ptr, (S32)(size * nmemb))) /*Flawfinder: ignore*/
- {
- S32 read = file->getLastBytesRead();
- return read / size; /*Flawfinder: ignore*/
- }
- else
- {
- return 0;
- }
-}
-
-S32 cache_seek(void *datasource, ogg_int64_t offset, S32 whence)
-{
- LLFileSystem *file = (LLFileSystem *)datasource;
-
- // cache has 31-bit files
- if (offset > S32_MAX)
- {
- return -1;
- }
-
- S32 origin;
- switch (whence) {
- case SEEK_SET:
- origin = 0;
- break;
- case SEEK_END:
- origin = file->getSize();
- break;
- case SEEK_CUR:
- origin = -1;
- break;
- default:
- LL_ERRS("AudioEngine") << "Invalid whence argument to cache_seek" << LL_ENDL;
- return -1;
- }
-
- if (file->seek((S32)offset, origin))
- {
- return 0;
- }
- else
- {
- return -1;
- }
-}
-
-S32 cache_close (void *datasource)
-{
- LLFileSystem *file = (LLFileSystem *)datasource;
- delete file;
- return 0;
-}
-
-long cache_tell (void *datasource)
-{
- LLFileSystem *file = (LLFileSystem *)datasource;
- return file->tell();
-}
-
-LLVorbisDecodeState::LLVorbisDecodeState(const LLUUID &uuid, const std::string &out_filename)
-{
- mDone = false;
- mValid = false;
- mBytesRead = -1;
- mUUID = uuid;
- mInFilep = NULL;
- mCurrentSection = 0;
- mOutFilename = out_filename;
- mFileHandle = LLLFSThread::nullHandle();
-
- // No default value for mVF, it's an ogg structure?
- // Hey, let's zero it anyway, for predictability.
- memset(&mVF, 0, sizeof(mVF));
-}
-
-LLVorbisDecodeState::~LLVorbisDecodeState()
-{
- if (!mDone)
- {
- delete mInFilep;
- mInFilep = NULL;
- }
-}
-
-
-bool LLVorbisDecodeState::initDecode()
-{
- ov_callbacks cache_callbacks;
- cache_callbacks.read_func = cache_read;
- cache_callbacks.seek_func = cache_seek;
- cache_callbacks.close_func = cache_close;
- cache_callbacks.tell_func = cache_tell;
-
- LL_DEBUGS("AudioEngine") << "Initing decode from vfile: " << mUUID << LL_ENDL;
-
- mInFilep = new LLFileSystem(mUUID, LLAssetType::AT_SOUND);
- if (!mInFilep || !mInFilep->getSize())
- {
- LL_WARNS("AudioEngine") << "unable to open vorbis source vfile for reading" << LL_ENDL;
- delete mInFilep;
- mInFilep = NULL;
- return false;
- }
-
- S32 r = ov_open_callbacks(mInFilep, &mVF, NULL, 0, cache_callbacks);
- if(r < 0)
- {
- LL_WARNS("AudioEngine") << r << " Input to vorbis decode does not appear to be an Ogg bitstream: " << mUUID << LL_ENDL;
- return(false);
- }
-
- S32 sample_count = (S32)ov_pcm_total(&mVF, -1);
- size_t size_guess = (size_t)sample_count;
- vorbis_info* vi = ov_info(&mVF, -1);
- size_guess *= (vi? vi->channels : 1);
- size_guess *= 2;
- size_guess += 2048;
-
- bool abort_decode = false;
-
- if (vi)
- {
- if( vi->channels < 1 || vi->channels > LLVORBIS_CLIP_MAX_CHANNELS )
- {
- abort_decode = true;
- LL_WARNS("AudioEngine") << "Bad channel count: " << vi->channels << LL_ENDL;
- }
- }
- else // !vi
- {
- abort_decode = true;
- LL_WARNS("AudioEngine") << "No default bitstream found" << LL_ENDL;
- }
-
- if( (size_t)sample_count > LLVORBIS_CLIP_REJECT_SAMPLES ||
- (size_t)sample_count <= 0)
- {
- abort_decode = true;
- LL_WARNS("AudioEngine") << "Illegal sample count: " << sample_count << LL_ENDL;
- }
-
- if( size_guess > LLVORBIS_CLIP_REJECT_SIZE )
- {
- abort_decode = true;
- LL_WARNS("AudioEngine") << "Illegal sample size: " << size_guess << LL_ENDL;
- }
-
- if( abort_decode )
- {
- LL_WARNS("AudioEngine") << "Canceling initDecode. Bad asset: " << mUUID << LL_ENDL;
- vorbis_comment* comment = ov_comment(&mVF,-1);
- if (comment && comment->vendor)
- {
- LL_WARNS("AudioEngine") << "Bad asset encoded by: " << comment->vendor << LL_ENDL;
- }
- delete mInFilep;
- mInFilep = NULL;
- return false;
- }
-
- try
- {
- mWAVBuffer.reserve(size_guess);
- mWAVBuffer.resize(WAV_HEADER_SIZE);
- }
- catch (std::bad_alloc&)
- {
- LL_WARNS("AudioEngine") << "Out of memory when trying to alloc buffer: " << size_guess << LL_ENDL;
- delete mInFilep;
- mInFilep = NULL;
- return false;
- }
-
- {
- // write the .wav format header
- //"RIFF"
- mWAVBuffer[0] = 0x52;
- mWAVBuffer[1] = 0x49;
- mWAVBuffer[2] = 0x46;
- mWAVBuffer[3] = 0x46;
-
- // length = datalen + 36 (to be filled in later)
- mWAVBuffer[4] = 0x00;
- mWAVBuffer[5] = 0x00;
- mWAVBuffer[6] = 0x00;
- mWAVBuffer[7] = 0x00;
-
- //"WAVE"
- mWAVBuffer[8] = 0x57;
- mWAVBuffer[9] = 0x41;
- mWAVBuffer[10] = 0x56;
- mWAVBuffer[11] = 0x45;
-
- // "fmt "
- mWAVBuffer[12] = 0x66;
- mWAVBuffer[13] = 0x6D;
- mWAVBuffer[14] = 0x74;
- mWAVBuffer[15] = 0x20;
-
- // chunk size = 16
- mWAVBuffer[16] = 0x10;
- mWAVBuffer[17] = 0x00;
- mWAVBuffer[18] = 0x00;
- mWAVBuffer[19] = 0x00;
-
- // format (1 = PCM)
- mWAVBuffer[20] = 0x01;
- mWAVBuffer[21] = 0x00;
-
- // number of channels
- mWAVBuffer[22] = 0x01;
- mWAVBuffer[23] = 0x00;
-
- // samples per second
- mWAVBuffer[24] = 0x44;
- mWAVBuffer[25] = 0xAC;
- mWAVBuffer[26] = 0x00;
- mWAVBuffer[27] = 0x00;
-
- // average bytes per second
- mWAVBuffer[28] = 0x88;
- mWAVBuffer[29] = 0x58;
- mWAVBuffer[30] = 0x01;
- mWAVBuffer[31] = 0x00;
-
- // bytes to output at a single time
- mWAVBuffer[32] = 0x02;
- mWAVBuffer[33] = 0x00;
-
- // 16 bits per sample
- mWAVBuffer[34] = 0x10;
- mWAVBuffer[35] = 0x00;
-
- // "data"
- mWAVBuffer[36] = 0x64;
- mWAVBuffer[37] = 0x61;
- mWAVBuffer[38] = 0x74;
- mWAVBuffer[39] = 0x61;
-
- // these are the length of the data chunk, to be filled in later
- mWAVBuffer[40] = 0x00;
- mWAVBuffer[41] = 0x00;
- mWAVBuffer[42] = 0x00;
- mWAVBuffer[43] = 0x00;
- }
-
- //{
- //char **ptr=ov_comment(&mVF,-1)->user_comments;
-// vorbis_info *vi=ov_info(&vf,-1);
- //while(*ptr){
- // fprintf(stderr,"%s\n",*ptr);
- // ++ptr;
- //}
-// fprintf(stderr,"\nBitstream is %d channel, %ldHz\n",vi->channels,vi->rate);
-// fprintf(stderr,"\nDecoded length: %ld samples\n", (long)ov_pcm_total(&vf,-1));
-// fprintf(stderr,"Encoded by: %s\n\n",ov_comment(&vf,-1)->vendor);
- //}
- return true;
-}
-
-bool LLVorbisDecodeState::decodeSection()
-{
- if (!mInFilep)
- {
- LL_WARNS("AudioEngine") << "No cache file to decode in vorbis!" << LL_ENDL;
- return true;
- }
- if (mDone)
- {
-// LL_WARNS("AudioEngine") << "Already done with decode, aborting!" << LL_ENDL;
- return true;
- }
- char pcmout[4096]; /*Flawfinder: ignore*/
-
- bool eof = false;
- long ret=ov_read(&mVF, pcmout, sizeof(pcmout), 0, 2, 1, &mCurrentSection);
- if (ret == 0)
- {
- /* EOF */
- eof = true;
- mDone = true;
- mValid = true;
-// LL_INFOS("AudioEngine") << "Vorbis EOF" << LL_ENDL;
- }
- else if (ret < 0)
- {
- /* error in the stream. Not a problem, just reporting it in
- case we (the app) cares. In this case, we don't. */
-
- LL_WARNS("AudioEngine") << "BAD vorbis decode in decodeSection." << LL_ENDL;
-
- mValid = false;
- mDone = true;
- // We're done, return true.
- return true;
- }
- else
- {
-// LL_INFOS("AudioEngine") << "Vorbis read " << ret << "bytes" << LL_ENDL;
- /* we don't bother dealing with sample rate changes, etc, but.
- you'll have to*/
- std::copy(pcmout, pcmout+ret, std::back_inserter(mWAVBuffer));
- }
- return eof;
-}
-
-bool LLVorbisDecodeState::finishDecode()
-{
- if (!isValid())
- {
- LL_WARNS("AudioEngine") << "Bogus vorbis decode state for " << getUUID() << ", aborting!" << LL_ENDL;
- return true; // We've finished
- }
-
- if (mFileHandle == LLLFSThread::nullHandle())
- {
- ov_clear(&mVF);
-
- // write "data" chunk length, in little-endian format
- S32 data_length = mWAVBuffer.size() - WAV_HEADER_SIZE;
- mWAVBuffer[40] = (data_length) & 0x000000FF;
- mWAVBuffer[41] = (data_length >> 8) & 0x000000FF;
- mWAVBuffer[42] = (data_length >> 16) & 0x000000FF;
- mWAVBuffer[43] = (data_length >> 24) & 0x000000FF;
- // write overall "RIFF" length, in little-endian format
- data_length += 36;
- mWAVBuffer[4] = (data_length) & 0x000000FF;
- mWAVBuffer[5] = (data_length >> 8) & 0x000000FF;
- mWAVBuffer[6] = (data_length >> 16) & 0x000000FF;
- mWAVBuffer[7] = (data_length >> 24) & 0x000000FF;
-
- //
- // FUDGECAKES!!! Vorbis encode/decode messes up loop point transitions (pop)
- // do a cheap-and-cheesy crossfade
- //
- {
- S16 *samplep;
- S32 i;
- S32 fade_length;
- char pcmout[4096]; /*Flawfinder: ignore*/
-
- fade_length = llmin((S32)128,(S32)(data_length-36)/8);
- if((S32)mWAVBuffer.size() >= (WAV_HEADER_SIZE + 2* fade_length))
- {
- memcpy(pcmout, &mWAVBuffer[WAV_HEADER_SIZE], (2 * fade_length)); /*Flawfinder: ignore*/
- }
- llendianswizzle(&pcmout, 2, fade_length);
-
- samplep = (S16 *)pcmout;
- for (i = 0 ;i < fade_length; i++)
- {
- *samplep = llfloor((F32)*samplep * ((F32)i/(F32)fade_length));
- samplep++;
- }
-
- llendianswizzle(&pcmout, 2, fade_length);
- if((WAV_HEADER_SIZE+(2 * fade_length)) < (S32)mWAVBuffer.size())
- {
- memcpy(&mWAVBuffer[WAV_HEADER_SIZE], pcmout, (2 * fade_length)); /*Flawfinder: ignore*/
- }
- S32 near_end = mWAVBuffer.size() - (2 * fade_length);
- if ((S32)mWAVBuffer.size() >= ( near_end + 2* fade_length))
- {
- memcpy(pcmout, &mWAVBuffer[near_end], (2 * fade_length)); /*Flawfinder: ignore*/
- }
- llendianswizzle(&pcmout, 2, fade_length);
-
- samplep = (S16 *)pcmout;
- for (i = fade_length-1 ; i >= 0; i--)
- {
- *samplep = llfloor((F32)*samplep * ((F32)i/(F32)fade_length));
- samplep++;
- }
-
- llendianswizzle(&pcmout, 2, fade_length);
- if (near_end + (2 * fade_length) < (S32)mWAVBuffer.size())
- {
- memcpy(&mWAVBuffer[near_end], pcmout, (2 * fade_length));/*Flawfinder: ignore*/
- }
- }
-
- if (36 == data_length)
- {
- LL_WARNS("AudioEngine") << "BAD Vorbis decode in finishDecode!" << LL_ENDL;
- mValid = false;
- return true; // we've finished
- }
- mBytesRead = -1;
- mFileHandle = LLLFSThread::sLocal->write(mOutFilename, &mWAVBuffer[0], 0, mWAVBuffer.size(),
- new WriteResponder(this));
- }
-
- if (mFileHandle != LLLFSThread::nullHandle())
- {
- if (mBytesRead >= 0)
- {
- if (mBytesRead == 0)
- {
- LL_WARNS("AudioEngine") << "Unable to write file in LLVorbisDecodeState::finishDecode" << LL_ENDL;
- mValid = false;
- return true; // we've finished
- }
- }
- else
- {
- return false; // not done
- }
- }
-
- mDone = true;
-
- LL_DEBUGS("AudioEngine") << "Finished decode for " << getUUID() << LL_ENDL;
-
- return true;
-}
-
-void LLVorbisDecodeState::flushBadFile()
-{
- if (mInFilep)
- {
- LL_WARNS("AudioEngine") << "Flushing bad vorbis file from cache for " << mUUID << LL_ENDL;
- mInFilep->remove();
- }
-}
-
-//////////////////////////////////////////////////////////////////////////////
-
-class LLAudioDecodeMgr::Impl
-{
- friend class LLAudioDecodeMgr;
- Impl();
- public:
-
- void processQueue();
-
- void startMoreDecodes();
- void enqueueFinishAudio(const LLUUID &decode_id, LLPointer<LLVorbisDecodeState>& decode_state);
- void checkDecodesFinished();
-
- protected:
- std::deque<LLUUID> mDecodeQueue;
- std::map<LLUUID, LLPointer<LLVorbisDecodeState>> mDecodes;
-};
-
-LLAudioDecodeMgr::Impl::Impl()
-{
-}
-
-// Returns the in-progress decode_state, which may be an empty LLPointer if
-// there was an error and there is no more work to be done.
-LLPointer<LLVorbisDecodeState> beginDecodingAndWritingAudio(const LLUUID &decode_id);
-
-// Return true if finished
-bool tryFinishAudio(const LLUUID &decode_id, LLPointer<LLVorbisDecodeState> decode_state);
-
-void LLAudioDecodeMgr::Impl::processQueue()
-{
- // First, check if any audio from in-progress decodes are ready to play. If
- // so, mark them ready for playback (or errored, in case of error).
- checkDecodesFinished();
-
- // Second, start as many decodes from the queue as permitted
- startMoreDecodes();
-}
-
-void LLAudioDecodeMgr::Impl::startMoreDecodes()
-{
- llassert_always(gAudiop);
-
- LL::WorkQueue::ptr_t main_queue = LL::WorkQueue::getInstance("mainloop");
- // *NOTE: main_queue->postTo casts this refcounted smart pointer to a weak
- // pointer
- LL::WorkQueue::ptr_t general_queue = LL::WorkQueue::getInstance("General");
- const LL::ThreadPool::ptr_t general_thread_pool = LL::ThreadPool::getInstance("General");
- llassert_always(main_queue);
- llassert_always(general_queue);
- llassert_always(general_thread_pool);
- // Set max decodes to double the thread count of the general work queue.
- // This ensures the general work queue is full, but prevents theoretical
- // buildup of buffers in memory due to disk writes once the
- // LLVorbisDecodeState leaves the worker thread (see
- // LLLFSThread::sLocal->write). This is probably as fast as we can get it
- // without modifying/removing LLVorbisDecodeState, at which point we should
- // consider decoding the audio during the asset download process.
- // -Cosmic,2022-05-11
- const size_t max_decodes = general_thread_pool->getWidth() * 2;
-
- while (!mDecodeQueue.empty() && mDecodes.size() < max_decodes)
- {
- const LLUUID decode_id = mDecodeQueue.front();
- mDecodeQueue.pop_front();
-
- // Don't decode the same file twice
- if (mDecodes.find(decode_id) != mDecodes.end())
- {
- continue;
- }
- if (gAudiop->hasDecodedFile(decode_id))
- {
- continue;
- }
-
- // Kick off a decode
- mDecodes[decode_id] = LLPointer<LLVorbisDecodeState>(NULL);
- bool posted = main_queue->postTo(
- general_queue,
- [decode_id]() // Work done on general queue
- {
- LLPointer<LLVorbisDecodeState> decode_state = beginDecodingAndWritingAudio(decode_id);
-
- if (!decode_state)
- {
- // Audio decode has errored
- return decode_state;
- }
-
- // Disk write of decoded audio is now in progress off-thread
- return decode_state;
- },
- [decode_id, this](LLPointer<LLVorbisDecodeState> decode_state) // Callback to main thread
- mutable {
- if (!gAudiop)
- {
- // There is no LLAudioEngine anymore. This might happen if
- // an audio decode is enqueued just before shutdown.
- return;
- }
-
- // At this point, we can be certain that the pointer to "this"
- // is valid because the lifetime of "this" is dependent upon
- // the lifetime of gAudiop.
-
- enqueueFinishAudio(decode_id, decode_state);
- });
- if (! posted)
- {
- // Shutdown
- // Consider making processQueue() do a cleanup instead
- // of starting more decodes
- LL_WARNS() << "Tried to start decoding on shutdown" << LL_ENDL;
- }
- }
-}
-
-LLPointer<LLVorbisDecodeState> beginDecodingAndWritingAudio(const LLUUID &decode_id)
-{
- LL_PROFILE_ZONE_SCOPED_CATEGORY_MEDIA;
-
- LL_DEBUGS() << "Decoding " << decode_id << " from audio queue!" << LL_ENDL;
-
- std::string d_path = gDirUtilp->getExpandedFilename(LL_PATH_CACHE, decode_id.asString()) + ".dsf";
- LLPointer<LLVorbisDecodeState> decode_state = new LLVorbisDecodeState(decode_id, d_path);
-
- if (!decode_state->initDecode())
- {
- return NULL;
- }
-
- // Decode in a loop until we're done
- while (!decode_state->decodeSection())
- {
- // decodeSection does all of the work above
- }
-
- if (!decode_state->isDone())
- {
- // Decode stopped early, or something bad happened to the file
- // during decoding.
- LL_WARNS("AudioEngine") << decode_id << " has invalid vorbis data or decode has been canceled, aborting decode" << LL_ENDL;
- decode_state->flushBadFile();
- return NULL;
- }
-
- if (!decode_state->isValid())
- {
- // We had an error when decoding, abort.
- LL_WARNS("AudioEngine") << decode_id << " has invalid vorbis data, aborting decode" << LL_ENDL;
- decode_state->flushBadFile();
- return NULL;
- }
-
- // Kick off the writing of the decoded audio to the disk cache.
- // The receiving thread can then cheaply call finishDecode() again to check
- // if writing has finished. Someone has to hold on to the refcounted
- // decode_state to prevent it from getting destroyed during write.
- decode_state->finishDecode();
-
- return decode_state;
-}
-
-void LLAudioDecodeMgr::Impl::enqueueFinishAudio(const LLUUID &decode_id, LLPointer<LLVorbisDecodeState>& decode_state)
-{
- // Assumed fast
- if (tryFinishAudio(decode_id, decode_state))
- {
- // Done early!
- auto decode_iter = mDecodes.find(decode_id);
- llassert(decode_iter != mDecodes.end());
- mDecodes.erase(decode_iter);
- return;
- }
-
- // Not done yet... enqueue it
- mDecodes[decode_id] = decode_state;
-}
-
-void LLAudioDecodeMgr::Impl::checkDecodesFinished()
-{
- auto decode_iter = mDecodes.begin();
- while (decode_iter != mDecodes.end())
- {
- const LLUUID& decode_id = decode_iter->first;
- const LLPointer<LLVorbisDecodeState>& decode_state = decode_iter->second;
- if (tryFinishAudio(decode_id, decode_state))
- {
- decode_iter = mDecodes.erase(decode_iter);
- }
- else
- {
- ++decode_iter;
- }
- }
-}
-
-bool tryFinishAudio(const LLUUID &decode_id, LLPointer<LLVorbisDecodeState> decode_state)
-{
- // decode_state is a file write in progress unless finished is true
- bool finished = decode_state && decode_state->finishDecode();
- if (!finished)
- {
- return false;
- }
-
- llassert_always(gAudiop);
-
- LLAudioData *adp = gAudiop->getAudioData(decode_id);
- if (!adp)
- {
- LL_WARNS("AudioEngine") << "Missing LLAudioData for decode of " << decode_id << LL_ENDL;
- return true;
- }
-
- bool valid = decode_state && decode_state->isValid();
- // Mark current decode finished regardless of success or failure
- adp->setHasCompletedDecode(true);
- // Flip flags for decoded data
- adp->setHasDecodeFailed(!valid);
- adp->setHasDecodedData(valid);
- // When finished decoding, there will also be a decoded wav file cached on
- // disk with the .dsf extension
- if (valid)
- {
- adp->setHasWAVLoadFailed(false);
- }
-
- return true;
-}
-
-//////////////////////////////////////////////////////////////////////////////
-
-LLAudioDecodeMgr::LLAudioDecodeMgr()
-{
- mImpl = new Impl();
-}
-
-LLAudioDecodeMgr::~LLAudioDecodeMgr()
-{
- delete mImpl;
- mImpl = nullptr;
-}
-
-void LLAudioDecodeMgr::processQueue()
-{
- mImpl->processQueue();
-}
-
-bool LLAudioDecodeMgr::addDecodeRequest(const LLUUID &uuid)
-{
- if (gAudiop && gAudiop->hasDecodedFile(uuid))
- {
- // Already have a decoded version, don't need to decode it.
- LL_DEBUGS("AudioEngine") << "addDecodeRequest for " << uuid << " has decoded file already" << LL_ENDL;
- return true;
- }
-
- if (gAssetStorage->hasLocalAsset(uuid, LLAssetType::AT_SOUND))
- {
- // Just put it on the decode queue.
- LL_DEBUGS("AudioEngine") << "addDecodeRequest for " << uuid << " has local asset file already" << LL_ENDL;
- mImpl->mDecodeQueue.push_back(uuid);
- return true;
- }
-
- LL_DEBUGS("AudioEngine") << "addDecodeRequest for " << uuid << " no file available" << LL_ENDL;
- return false;
-}
+/**
+ * @file llaudiodecodemgr.cpp
+ *
+ * $LicenseInfo:firstyear=2003&license=viewerlgpl$
+ * Second Life Viewer Source Code
+ * Copyright (C) 2010, Linden Research, Inc.
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation;
+ * version 2.1 of the License only.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with this library; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ *
+ * Linden Research, Inc., 945 Battery Street, San Francisco, CA 94111 USA
+ * $/LicenseInfo$
+ */
+
+#include "linden_common.h"
+
+#include "llaudiodecodemgr.h"
+
+#include "llaudioengine.h"
+#include "lllfsthread.h"
+#include "llfilesystem.h"
+#include "llstring.h"
+#include "lldir.h"
+#include "llendianswizzle.h"
+#include "llassetstorage.h"
+#include "llrefcount.h"
+#include "threadpool.h"
+#include "workqueue.h"
+
+#include "llvorbisencode.h"
+
+#include "vorbis/codec.h"
+#include "vorbis/vorbisfile.h"
+#include <iterator>
+#include <deque>
+
+extern LLAudioEngine *gAudiop;
+
+static const S32 WAV_HEADER_SIZE = 44;
+
+
+//////////////////////////////////////////////////////////////////////////////
+
+
+class LLVorbisDecodeState : public LLThreadSafeRefCount
+{
+public:
+ class WriteResponder : public LLLFSThread::Responder
+ {
+ public:
+ WriteResponder(LLVorbisDecodeState* decoder) : mDecoder(decoder) {}
+ ~WriteResponder() {}
+ void completed(S32 bytes)
+ {
+ mDecoder->ioComplete(bytes);
+ }
+ LLPointer<LLVorbisDecodeState> mDecoder;
+ };
+
+ LLVorbisDecodeState(const LLUUID &uuid, const std::string &out_filename);
+
+ bool initDecode();
+ bool decodeSection(); // Return true if done.
+ bool finishDecode();
+
+ void flushBadFile();
+
+ void ioComplete(S32 bytes) { mBytesRead = bytes; }
+ bool isValid() const { return mValid; }
+ bool isDone() const { return mDone; }
+ const LLUUID &getUUID() const { return mUUID; }
+
+protected:
+ virtual ~LLVorbisDecodeState();
+
+ bool mValid;
+ bool mDone;
+ LLAtomicS32 mBytesRead;
+ LLUUID mUUID;
+
+ std::vector<U8> mWAVBuffer;
+ std::string mOutFilename;
+ LLLFSThread::handle_t mFileHandle;
+
+ LLFileSystem *mInFilep;
+ OggVorbis_File mVF;
+ S32 mCurrentSection;
+};
+
+size_t cache_read(void *ptr, size_t size, size_t nmemb, void *datasource)
+{
+ LLFileSystem *file = (LLFileSystem *)datasource;
+
+ if (file->read((U8*)ptr, (S32)(size * nmemb))) /*Flawfinder: ignore*/
+ {
+ S32 read = file->getLastBytesRead();
+ return read / size; /*Flawfinder: ignore*/
+ }
+ else
+ {
+ return 0;
+ }
+}
+
+S32 cache_seek(void *datasource, ogg_int64_t offset, S32 whence)
+{
+ LLFileSystem *file = (LLFileSystem *)datasource;
+
+ // cache has 31-bit files
+ if (offset > S32_MAX)
+ {
+ return -1;
+ }
+
+ S32 origin;
+ switch (whence) {
+ case SEEK_SET:
+ origin = 0;
+ break;
+ case SEEK_END:
+ origin = file->getSize();
+ break;
+ case SEEK_CUR:
+ origin = -1;
+ break;
+ default:
+ LL_ERRS("AudioEngine") << "Invalid whence argument to cache_seek" << LL_ENDL;
+ return -1;
+ }
+
+ if (file->seek((S32)offset, origin))
+ {
+ return 0;
+ }
+ else
+ {
+ return -1;
+ }
+}
+
+S32 cache_close (void *datasource)
+{
+ LLFileSystem *file = (LLFileSystem *)datasource;
+ delete file;
+ return 0;
+}
+
+long cache_tell (void *datasource)
+{
+ LLFileSystem *file = (LLFileSystem *)datasource;
+ return file->tell();
+}
+
+LLVorbisDecodeState::LLVorbisDecodeState(const LLUUID &uuid, const std::string &out_filename)
+{
+ mDone = false;
+ mValid = false;
+ mBytesRead = -1;
+ mUUID = uuid;
+ mInFilep = NULL;
+ mCurrentSection = 0;
+ mOutFilename = out_filename;
+ mFileHandle = LLLFSThread::nullHandle();
+
+ // No default value for mVF, it's an ogg structure?
+ // Hey, let's zero it anyway, for predictability.
+ memset(&mVF, 0, sizeof(mVF));
+}
+
+LLVorbisDecodeState::~LLVorbisDecodeState()
+{
+ if (!mDone)
+ {
+ delete mInFilep;
+ mInFilep = NULL;
+ }
+}
+
+
+bool LLVorbisDecodeState::initDecode()
+{
+ ov_callbacks cache_callbacks;
+ cache_callbacks.read_func = cache_read;
+ cache_callbacks.seek_func = cache_seek;
+ cache_callbacks.close_func = cache_close;
+ cache_callbacks.tell_func = cache_tell;
+
+ LL_DEBUGS("AudioEngine") << "Initing decode from vfile: " << mUUID << LL_ENDL;
+
+ mInFilep = new LLFileSystem(mUUID, LLAssetType::AT_SOUND);
+ if (!mInFilep || !mInFilep->getSize())
+ {
+ LL_WARNS("AudioEngine") << "unable to open vorbis source vfile for reading" << LL_ENDL;
+ delete mInFilep;
+ mInFilep = NULL;
+ return false;
+ }
+
+ S32 r = ov_open_callbacks(mInFilep, &mVF, NULL, 0, cache_callbacks);
+ if(r < 0)
+ {
+ LL_WARNS("AudioEngine") << r << " Input to vorbis decode does not appear to be an Ogg bitstream: " << mUUID << LL_ENDL;
+ return(false);
+ }
+
+ S32 sample_count = (S32)ov_pcm_total(&mVF, -1);
+ size_t size_guess = (size_t)sample_count;
+ vorbis_info* vi = ov_info(&mVF, -1);
+ size_guess *= (vi? vi->channels : 1);
+ size_guess *= 2;
+ size_guess += 2048;
+
+ bool abort_decode = false;
+
+ if (vi)
+ {
+ if( vi->channels < 1 || vi->channels > LLVORBIS_CLIP_MAX_CHANNELS )
+ {
+ abort_decode = true;
+ LL_WARNS("AudioEngine") << "Bad channel count: " << vi->channels << LL_ENDL;
+ }
+ }
+ else // !vi
+ {
+ abort_decode = true;
+ LL_WARNS("AudioEngine") << "No default bitstream found" << LL_ENDL;
+ }
+
+ if( (size_t)sample_count > LLVORBIS_CLIP_REJECT_SAMPLES ||
+ (size_t)sample_count <= 0)
+ {
+ abort_decode = true;
+ LL_WARNS("AudioEngine") << "Illegal sample count: " << sample_count << LL_ENDL;
+ }
+
+ if( size_guess > LLVORBIS_CLIP_REJECT_SIZE )
+ {
+ abort_decode = true;
+ LL_WARNS("AudioEngine") << "Illegal sample size: " << size_guess << LL_ENDL;
+ }
+
+ if( abort_decode )
+ {
+ LL_WARNS("AudioEngine") << "Canceling initDecode. Bad asset: " << mUUID << LL_ENDL;
+ vorbis_comment* comment = ov_comment(&mVF,-1);
+ if (comment && comment->vendor)
+ {
+ LL_WARNS("AudioEngine") << "Bad asset encoded by: " << comment->vendor << LL_ENDL;
+ }
+ delete mInFilep;
+ mInFilep = NULL;
+ return false;
+ }
+
+ try
+ {
+ mWAVBuffer.reserve(size_guess);
+ mWAVBuffer.resize(WAV_HEADER_SIZE);
+ }
+ catch (std::bad_alloc&)
+ {
+ LL_WARNS("AudioEngine") << "Out of memory when trying to alloc buffer: " << size_guess << LL_ENDL;
+ delete mInFilep;
+ mInFilep = NULL;
+ return false;
+ }
+
+ {
+ // write the .wav format header
+ //"RIFF"
+ mWAVBuffer[0] = 0x52;
+ mWAVBuffer[1] = 0x49;
+ mWAVBuffer[2] = 0x46;
+ mWAVBuffer[3] = 0x46;
+
+ // length = datalen + 36 (to be filled in later)
+ mWAVBuffer[4] = 0x00;
+ mWAVBuffer[5] = 0x00;
+ mWAVBuffer[6] = 0x00;
+ mWAVBuffer[7] = 0x00;
+
+ //"WAVE"
+ mWAVBuffer[8] = 0x57;
+ mWAVBuffer[9] = 0x41;
+ mWAVBuffer[10] = 0x56;
+ mWAVBuffer[11] = 0x45;
+
+ // "fmt "
+ mWAVBuffer[12] = 0x66;
+ mWAVBuffer[13] = 0x6D;
+ mWAVBuffer[14] = 0x74;
+ mWAVBuffer[15] = 0x20;
+
+ // chunk size = 16
+ mWAVBuffer[16] = 0x10;
+ mWAVBuffer[17] = 0x00;
+ mWAVBuffer[18] = 0x00;
+ mWAVBuffer[19] = 0x00;
+
+ // format (1 = PCM)
+ mWAVBuffer[20] = 0x01;
+ mWAVBuffer[21] = 0x00;
+
+ // number of channels
+ mWAVBuffer[22] = 0x01;
+ mWAVBuffer[23] = 0x00;
+
+ // samples per second
+ mWAVBuffer[24] = 0x44;
+ mWAVBuffer[25] = 0xAC;
+ mWAVBuffer[26] = 0x00;
+ mWAVBuffer[27] = 0x00;
+
+ // average bytes per second
+ mWAVBuffer[28] = 0x88;
+ mWAVBuffer[29] = 0x58;
+ mWAVBuffer[30] = 0x01;
+ mWAVBuffer[31] = 0x00;
+
+ // bytes to output at a single time
+ mWAVBuffer[32] = 0x02;
+ mWAVBuffer[33] = 0x00;
+
+ // 16 bits per sample
+ mWAVBuffer[34] = 0x10;
+ mWAVBuffer[35] = 0x00;
+
+ // "data"
+ mWAVBuffer[36] = 0x64;
+ mWAVBuffer[37] = 0x61;
+ mWAVBuffer[38] = 0x74;
+ mWAVBuffer[39] = 0x61;
+
+ // these are the length of the data chunk, to be filled in later
+ mWAVBuffer[40] = 0x00;
+ mWAVBuffer[41] = 0x00;
+ mWAVBuffer[42] = 0x00;
+ mWAVBuffer[43] = 0x00;
+ }
+
+ //{
+ //char **ptr=ov_comment(&mVF,-1)->user_comments;
+// vorbis_info *vi=ov_info(&vf,-1);
+ //while(*ptr){
+ // fprintf(stderr,"%s\n",*ptr);
+ // ++ptr;
+ //}
+// fprintf(stderr,"\nBitstream is %d channel, %ldHz\n",vi->channels,vi->rate);
+// fprintf(stderr,"\nDecoded length: %ld samples\n", (long)ov_pcm_total(&vf,-1));
+// fprintf(stderr,"Encoded by: %s\n\n",ov_comment(&vf,-1)->vendor);
+ //}
+ return true;
+}
+
+bool LLVorbisDecodeState::decodeSection()
+{
+ if (!mInFilep)
+ {
+ LL_WARNS("AudioEngine") << "No cache file to decode in vorbis!" << LL_ENDL;
+ return true;
+ }
+ if (mDone)
+ {
+// LL_WARNS("AudioEngine") << "Already done with decode, aborting!" << LL_ENDL;
+ return true;
+ }
+ char pcmout[4096]; /*Flawfinder: ignore*/
+
+ bool eof = false;
+ long ret=ov_read(&mVF, pcmout, sizeof(pcmout), 0, 2, 1, &mCurrentSection);
+ if (ret == 0)
+ {
+ /* EOF */
+ eof = true;
+ mDone = true;
+ mValid = true;
+// LL_INFOS("AudioEngine") << "Vorbis EOF" << LL_ENDL;
+ }
+ else if (ret < 0)
+ {
+ /* error in the stream. Not a problem, just reporting it in
+ case we (the app) cares. In this case, we don't. */
+
+ LL_WARNS("AudioEngine") << "BAD vorbis decode in decodeSection." << LL_ENDL;
+
+ mValid = false;
+ mDone = true;
+ // We're done, return true.
+ return true;
+ }
+ else
+ {
+// LL_INFOS("AudioEngine") << "Vorbis read " << ret << "bytes" << LL_ENDL;
+ /* we don't bother dealing with sample rate changes, etc, but.
+ you'll have to*/
+ std::copy(pcmout, pcmout+ret, std::back_inserter(mWAVBuffer));
+ }
+ return eof;
+}
+
+bool LLVorbisDecodeState::finishDecode()
+{
+ if (!isValid())
+ {
+ LL_WARNS("AudioEngine") << "Bogus vorbis decode state for " << getUUID() << ", aborting!" << LL_ENDL;
+ return true; // We've finished
+ }
+
+ if (mFileHandle == LLLFSThread::nullHandle())
+ {
+ ov_clear(&mVF);
+
+ // write "data" chunk length, in little-endian format
+ S32 data_length = mWAVBuffer.size() - WAV_HEADER_SIZE;
+ mWAVBuffer[40] = (data_length) & 0x000000FF;
+ mWAVBuffer[41] = (data_length >> 8) & 0x000000FF;
+ mWAVBuffer[42] = (data_length >> 16) & 0x000000FF;
+ mWAVBuffer[43] = (data_length >> 24) & 0x000000FF;
+ // write overall "RIFF" length, in little-endian format
+ data_length += 36;
+ mWAVBuffer[4] = (data_length) & 0x000000FF;
+ mWAVBuffer[5] = (data_length >> 8) & 0x000000FF;
+ mWAVBuffer[6] = (data_length >> 16) & 0x000000FF;
+ mWAVBuffer[7] = (data_length >> 24) & 0x000000FF;
+
+ //
+ // FUDGECAKES!!! Vorbis encode/decode messes up loop point transitions (pop)
+ // do a cheap-and-cheesy crossfade
+ //
+ {
+ S16 *samplep;
+ S32 i;
+ S32 fade_length;
+ char pcmout[4096]; /*Flawfinder: ignore*/
+
+ fade_length = llmin((S32)128,(S32)(data_length-36)/8);
+ if((S32)mWAVBuffer.size() >= (WAV_HEADER_SIZE + 2* fade_length))
+ {
+ memcpy(pcmout, &mWAVBuffer[WAV_HEADER_SIZE], (2 * fade_length)); /*Flawfinder: ignore*/
+ }
+ llendianswizzle(&pcmout, 2, fade_length);
+
+ samplep = (S16 *)pcmout;
+ for (i = 0 ;i < fade_length; i++)
+ {
+ *samplep = llfloor((F32)*samplep * ((F32)i/(F32)fade_length));
+ samplep++;
+ }
+
+ llendianswizzle(&pcmout, 2, fade_length);
+ if((WAV_HEADER_SIZE+(2 * fade_length)) < (S32)mWAVBuffer.size())
+ {
+ memcpy(&mWAVBuffer[WAV_HEADER_SIZE], pcmout, (2 * fade_length)); /*Flawfinder: ignore*/
+ }
+ S32 near_end = mWAVBuffer.size() - (2 * fade_length);
+ if ((S32)mWAVBuffer.size() >= ( near_end + 2* fade_length))
+ {
+ memcpy(pcmout, &mWAVBuffer[near_end], (2 * fade_length)); /*Flawfinder: ignore*/
+ }
+ llendianswizzle(&pcmout, 2, fade_length);
+
+ samplep = (S16 *)pcmout;
+ for (i = fade_length-1 ; i >= 0; i--)
+ {
+ *samplep = llfloor((F32)*samplep * ((F32)i/(F32)fade_length));
+ samplep++;
+ }
+
+ llendianswizzle(&pcmout, 2, fade_length);
+ if (near_end + (2 * fade_length) < (S32)mWAVBuffer.size())
+ {
+ memcpy(&mWAVBuffer[near_end], pcmout, (2 * fade_length));/*Flawfinder: ignore*/
+ }
+ }
+
+ if (36 == data_length)
+ {
+ LL_WARNS("AudioEngine") << "BAD Vorbis decode in finishDecode!" << LL_ENDL;
+ mValid = false;
+ return true; // we've finished
+ }
+ mBytesRead = -1;
+ mFileHandle = LLLFSThread::sLocal->write(mOutFilename, &mWAVBuffer[0], 0, mWAVBuffer.size(),
+ new WriteResponder(this));
+ }
+
+ if (mFileHandle != LLLFSThread::nullHandle())
+ {
+ if (mBytesRead >= 0)
+ {
+ if (mBytesRead == 0)
+ {
+ LL_WARNS("AudioEngine") << "Unable to write file in LLVorbisDecodeState::finishDecode" << LL_ENDL;
+ mValid = false;
+ return true; // we've finished
+ }
+ }
+ else
+ {
+ return false; // not done
+ }
+ }
+
+ mDone = true;
+
+ LL_DEBUGS("AudioEngine") << "Finished decode for " << getUUID() << LL_ENDL;
+
+ return true;
+}
+
+void LLVorbisDecodeState::flushBadFile()
+{
+ if (mInFilep)
+ {
+ LL_WARNS("AudioEngine") << "Flushing bad vorbis file from cache for " << mUUID << LL_ENDL;
+ mInFilep->remove();
+ }
+}
+
+//////////////////////////////////////////////////////////////////////////////
+
+class LLAudioDecodeMgr::Impl
+{
+ friend class LLAudioDecodeMgr;
+ Impl();
+ public:
+
+ void processQueue();
+
+ void startMoreDecodes();
+ void enqueueFinishAudio(const LLUUID &decode_id, LLPointer<LLVorbisDecodeState>& decode_state);
+ void checkDecodesFinished();
+
+ protected:
+ std::deque<LLUUID> mDecodeQueue;
+ std::map<LLUUID, LLPointer<LLVorbisDecodeState>> mDecodes;
+};
+
+LLAudioDecodeMgr::Impl::Impl()
+{
+}
+
+// Returns the in-progress decode_state, which may be an empty LLPointer if
+// there was an error and there is no more work to be done.
+LLPointer<LLVorbisDecodeState> beginDecodingAndWritingAudio(const LLUUID &decode_id);
+
+// Return true if finished
+bool tryFinishAudio(const LLUUID &decode_id, LLPointer<LLVorbisDecodeState> decode_state);
+
+void LLAudioDecodeMgr::Impl::processQueue()
+{
+ // First, check if any audio from in-progress decodes are ready to play. If
+ // so, mark them ready for playback (or errored, in case of error).
+ checkDecodesFinished();
+
+ // Second, start as many decodes from the queue as permitted
+ startMoreDecodes();
+}
+
+void LLAudioDecodeMgr::Impl::startMoreDecodes()
+{
+ llassert_always(gAudiop);
+
+ LL::WorkQueue::ptr_t main_queue = LL::WorkQueue::getInstance("mainloop");
+ // *NOTE: main_queue->postTo casts this refcounted smart pointer to a weak
+ // pointer
+ LL::WorkQueue::ptr_t general_queue = LL::WorkQueue::getInstance("General");
+ const LL::ThreadPool::ptr_t general_thread_pool = LL::ThreadPool::getInstance("General");
+ llassert_always(main_queue);
+ llassert_always(general_queue);
+ llassert_always(general_thread_pool);
+ // Set max decodes to double the thread count of the general work queue.
+ // This ensures the general work queue is full, but prevents theoretical
+ // buildup of buffers in memory due to disk writes once the
+ // LLVorbisDecodeState leaves the worker thread (see
+ // LLLFSThread::sLocal->write). This is probably as fast as we can get it
+ // without modifying/removing LLVorbisDecodeState, at which point we should
+ // consider decoding the audio during the asset download process.
+ // -Cosmic,2022-05-11
+ const size_t max_decodes = general_thread_pool->getWidth() * 2;
+
+ while (!mDecodeQueue.empty() && mDecodes.size() < max_decodes)
+ {
+ const LLUUID decode_id = mDecodeQueue.front();
+ mDecodeQueue.pop_front();
+
+ // Don't decode the same file twice
+ if (mDecodes.find(decode_id) != mDecodes.end())
+ {
+ continue;
+ }
+ if (gAudiop->hasDecodedFile(decode_id))
+ {
+ continue;
+ }
+
+ // Kick off a decode
+ mDecodes[decode_id] = LLPointer<LLVorbisDecodeState>(NULL);
+ bool posted = main_queue->postTo(
+ general_queue,
+ [decode_id]() // Work done on general queue
+ {
+ LLPointer<LLVorbisDecodeState> decode_state = beginDecodingAndWritingAudio(decode_id);
+
+ if (!decode_state)
+ {
+ // Audio decode has errored
+ return decode_state;
+ }
+
+ // Disk write of decoded audio is now in progress off-thread
+ return decode_state;
+ },
+ [decode_id, this](LLPointer<LLVorbisDecodeState> decode_state) // Callback to main thread
+ mutable {
+ if (!gAudiop)
+ {
+ // There is no LLAudioEngine anymore. This might happen if
+ // an audio decode is enqueued just before shutdown.
+ return;
+ }
+
+ // At this point, we can be certain that the pointer to "this"
+ // is valid because the lifetime of "this" is dependent upon
+ // the lifetime of gAudiop.
+
+ enqueueFinishAudio(decode_id, decode_state);
+ });
+ if (! posted)
+ {
+ // Shutdown
+ // Consider making processQueue() do a cleanup instead
+ // of starting more decodes
+ LL_WARNS() << "Tried to start decoding on shutdown" << LL_ENDL;
+ }
+ }
+}
+
+LLPointer<LLVorbisDecodeState> beginDecodingAndWritingAudio(const LLUUID &decode_id)
+{
+ LL_PROFILE_ZONE_SCOPED_CATEGORY_MEDIA;
+
+ LL_DEBUGS() << "Decoding " << decode_id << " from audio queue!" << LL_ENDL;
+
+ std::string d_path = gDirUtilp->getExpandedFilename(LL_PATH_CACHE, decode_id.asString()) + ".dsf";
+ LLPointer<LLVorbisDecodeState> decode_state = new LLVorbisDecodeState(decode_id, d_path);
+
+ if (!decode_state->initDecode())
+ {
+ return NULL;
+ }
+
+ // Decode in a loop until we're done
+ while (!decode_state->decodeSection())
+ {
+ // decodeSection does all of the work above
+ }
+
+ if (!decode_state->isDone())
+ {
+ // Decode stopped early, or something bad happened to the file
+ // during decoding.
+ LL_WARNS("AudioEngine") << decode_id << " has invalid vorbis data or decode has been canceled, aborting decode" << LL_ENDL;
+ decode_state->flushBadFile();
+ return NULL;
+ }
+
+ if (!decode_state->isValid())
+ {
+ // We had an error when decoding, abort.
+ LL_WARNS("AudioEngine") << decode_id << " has invalid vorbis data, aborting decode" << LL_ENDL;
+ decode_state->flushBadFile();
+ return NULL;
+ }
+
+ // Kick off the writing of the decoded audio to the disk cache.
+ // The receiving thread can then cheaply call finishDecode() again to check
+ // if writing has finished. Someone has to hold on to the refcounted
+ // decode_state to prevent it from getting destroyed during write.
+ decode_state->finishDecode();
+
+ return decode_state;
+}
+
+void LLAudioDecodeMgr::Impl::enqueueFinishAudio(const LLUUID &decode_id, LLPointer<LLVorbisDecodeState>& decode_state)
+{
+ // Assumed fast
+ if (tryFinishAudio(decode_id, decode_state))
+ {
+ // Done early!
+ auto decode_iter = mDecodes.find(decode_id);
+ llassert(decode_iter != mDecodes.end());
+ mDecodes.erase(decode_iter);
+ return;
+ }
+
+ // Not done yet... enqueue it
+ mDecodes[decode_id] = decode_state;
+}
+
+void LLAudioDecodeMgr::Impl::checkDecodesFinished()
+{
+ auto decode_iter = mDecodes.begin();
+ while (decode_iter != mDecodes.end())
+ {
+ const LLUUID& decode_id = decode_iter->first;
+ const LLPointer<LLVorbisDecodeState>& decode_state = decode_iter->second;
+ if (tryFinishAudio(decode_id, decode_state))
+ {
+ decode_iter = mDecodes.erase(decode_iter);
+ }
+ else
+ {
+ ++decode_iter;
+ }
+ }
+}
+
+bool tryFinishAudio(const LLUUID &decode_id, LLPointer<LLVorbisDecodeState> decode_state)
+{
+ // decode_state is a file write in progress unless finished is true
+ bool finished = decode_state && decode_state->finishDecode();
+ if (!finished)
+ {
+ return false;
+ }
+
+ llassert_always(gAudiop);
+
+ LLAudioData *adp = gAudiop->getAudioData(decode_id);
+ if (!adp)
+ {
+ LL_WARNS("AudioEngine") << "Missing LLAudioData for decode of " << decode_id << LL_ENDL;
+ return true;
+ }
+
+ bool valid = decode_state && decode_state->isValid();
+ // Mark current decode finished regardless of success or failure
+ adp->setHasCompletedDecode(true);
+ // Flip flags for decoded data
+ adp->setHasDecodeFailed(!valid);
+ adp->setHasDecodedData(valid);
+ // When finished decoding, there will also be a decoded wav file cached on
+ // disk with the .dsf extension
+ if (valid)
+ {
+ adp->setHasWAVLoadFailed(false);
+ }
+
+ return true;
+}
+
+//////////////////////////////////////////////////////////////////////////////
+
+LLAudioDecodeMgr::LLAudioDecodeMgr()
+{
+ mImpl = new Impl();
+}
+
+LLAudioDecodeMgr::~LLAudioDecodeMgr()
+{
+ delete mImpl;
+ mImpl = nullptr;
+}
+
+void LLAudioDecodeMgr::processQueue()
+{
+ mImpl->processQueue();
+}
+
+bool LLAudioDecodeMgr::addDecodeRequest(const LLUUID &uuid)
+{
+ if (gAudiop && gAudiop->hasDecodedFile(uuid))
+ {
+ // Already have a decoded version, don't need to decode it.
+ LL_DEBUGS("AudioEngine") << "addDecodeRequest for " << uuid << " has decoded file already" << LL_ENDL;
+ return true;
+ }
+
+ if (gAssetStorage->hasLocalAsset(uuid, LLAssetType::AT_SOUND))
+ {
+ // Just put it on the decode queue.
+ LL_DEBUGS("AudioEngine") << "addDecodeRequest for " << uuid << " has local asset file already" << LL_ENDL;
+ mImpl->mDecodeQueue.push_back(uuid);
+ return true;
+ }
+
+ LL_DEBUGS("AudioEngine") << "addDecodeRequest for " << uuid << " no file available" << LL_ENDL;
+ return false;
+}
diff --git a/indra/llaudio/llaudiodecodemgr.h b/indra/llaudio/llaudiodecodemgr.h
index 17fa31ee53..79f8b8e92e 100644
--- a/indra/llaudio/llaudiodecodemgr.h
+++ b/indra/llaudio/llaudiodecodemgr.h
@@ -1,54 +1,54 @@
-/**
- * @file llaudiodecodemgr.h
- *
- * $LicenseInfo:firstyear=2003&license=viewerlgpl$
- * Second Life Viewer Source Code
- * Copyright (C) 2010, Linden Research, Inc.
- *
- * This library is free software; you can redistribute it and/or
- * modify it under the terms of the GNU Lesser General Public
- * License as published by the Free Software Foundation;
- * version 2.1 of the License only.
- *
- * This library is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
- * Lesser General Public License for more details.
- *
- * You should have received a copy of the GNU Lesser General Public
- * License along with this library; if not, write to the Free Software
- * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
- *
- * Linden Research, Inc., 945 Battery Street, San Francisco, CA 94111 USA
- * $/LicenseInfo$
- */
-
-#ifndef LL_LLAUDIODECODEMGR_H
-#define LL_LLAUDIODECODEMGR_H
-
-#include "stdtypes.h"
-
-#include "lluuid.h"
-
-#include "llassettype.h"
-#include "llframetimer.h"
-#include "llsingleton.h"
-
-template<class T> class LLPointer;
-class LLVorbisDecodeState;
-
-class LLAudioDecodeMgr : public LLSingleton<LLAudioDecodeMgr>
-{
- LLSINGLETON(LLAudioDecodeMgr);
- ~LLAudioDecodeMgr();
-public:
- void processQueue();
- bool addDecodeRequest(const LLUUID &uuid);
- void addAudioRequest(const LLUUID &uuid);
-
-protected:
- class Impl;
- Impl* mImpl;
-};
-
-#endif
+/**
+ * @file llaudiodecodemgr.h
+ *
+ * $LicenseInfo:firstyear=2003&license=viewerlgpl$
+ * Second Life Viewer Source Code
+ * Copyright (C) 2010, Linden Research, Inc.
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation;
+ * version 2.1 of the License only.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with this library; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ *
+ * Linden Research, Inc., 945 Battery Street, San Francisco, CA 94111 USA
+ * $/LicenseInfo$
+ */
+
+#ifndef LL_LLAUDIODECODEMGR_H
+#define LL_LLAUDIODECODEMGR_H
+
+#include "stdtypes.h"
+
+#include "lluuid.h"
+
+#include "llassettype.h"
+#include "llframetimer.h"
+#include "llsingleton.h"
+
+template<class T> class LLPointer;
+class LLVorbisDecodeState;
+
+class LLAudioDecodeMgr : public LLSingleton<LLAudioDecodeMgr>
+{
+ LLSINGLETON(LLAudioDecodeMgr);
+ ~LLAudioDecodeMgr();
+public:
+ void processQueue();
+ bool addDecodeRequest(const LLUUID &uuid);
+ void addAudioRequest(const LLUUID &uuid);
+
+protected:
+ class Impl;
+ Impl* mImpl;
+};
+
+#endif
diff --git a/indra/llaudio/llaudioengine.h b/indra/llaudio/llaudioengine.h
index 9949b8f337..a9a229c0a5 100755
--- a/indra/llaudio/llaudioengine.h
+++ b/indra/llaudio/llaudioengine.h
@@ -1,486 +1,486 @@
-/**
- * @file audioengine.h
- * @brief Definition of LLAudioEngine base class abstracting the audio support
- *
- * $LicenseInfo:firstyear=2000&license=viewerlgpl$
- * Second Life Viewer Source Code
- * Copyright (C) 2010, Linden Research, Inc.
- *
- * This library is free software; you can redistribute it and/or
- * modify it under the terms of the GNU Lesser General Public
- * License as published by the Free Software Foundation;
- * version 2.1 of the License only.
- *
- * This library is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
- * Lesser General Public License for more details.
- *
- * You should have received a copy of the GNU Lesser General Public
- * License along with this library; if not, write to the Free Software
- * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
- *
- * Linden Research, Inc., 945 Battery Street, San Francisco, CA 94111 USA
- * $/LicenseInfo$
- */
-
-
-#ifndef LL_AUDIOENGINE_H
-#define LL_AUDIOENGINE_H
-
-#include <list>
-#include <map>
-#include <array>
-
-#include "v3math.h"
-#include "v3dmath.h"
-#include "lltimer.h"
-#include "lluuid.h"
-#include "llframetimer.h"
-#include "llassettype.h"
-#include "llextendedstatus.h"
-
-#include "lllistener.h"
-
-const F32 LL_WIND_UPDATE_INTERVAL = 0.1f;
-const F32 LL_WIND_UNDERWATER_CENTER_FREQ = 20.f;
-
-const F32 ATTACHED_OBJECT_TIMEOUT = 5.0f;
-const F32 DEFAULT_MIN_DISTANCE = 2.0f;
-
-#define LL_MAX_AUDIO_CHANNELS 30
-#define LL_MAX_AUDIO_BUFFERS 40 // Some extra for preloading, maybe?
-
-class LLAudioSource;
-class LLAudioData;
-class LLAudioChannel;
-class LLAudioChannelOpenAL;
-class LLAudioBuffer;
-class LLStreamingAudioInterface;
-struct SoundData;
-
-//
-// LLAudioEngine definition
-//
-class LLAudioEngine
-{
- friend class LLAudioChannelOpenAL; // bleh. channel needs some listener methods.
-
-public:
- enum LLAudioType
- {
- AUDIO_TYPE_NONE = 0,
- AUDIO_TYPE_SFX = 1,
- AUDIO_TYPE_UI = 2,
- AUDIO_TYPE_AMBIENT = 3,
- AUDIO_TYPE_COUNT = 4 // last
- };
-
- enum LLAudioPlayState
- {
- // isInternetStreamPlaying() returns an *S32*, with
- // 0 = stopped, 1 = playing, 2 = paused.
- AUDIO_STOPPED = 0,
- AUDIO_PLAYING = 1,
- AUDIO_PAUSED = 2
- };
-
- LLAudioEngine();
- virtual ~LLAudioEngine();
-
- // initialization/startup/shutdown
- virtual bool init(void *userdata, const std::string &app_title);
- virtual std::string getDriverName(bool verbose) = 0;
- virtual LLStreamingAudioInterface *createDefaultStreamingAudioImpl() const = 0;
- virtual void shutdown();
-
- // Used by the mechanics of the engine
- //virtual void processQueue(const LLUUID &sound_guid);
- virtual void setListener(LLVector3 pos,LLVector3 vel,LLVector3 up,LLVector3 at);
- virtual void updateWind(LLVector3 direction, F32 camera_height_above_water) = 0;
- virtual void idle();
- virtual void updateChannels();
-
- //
- // "End user" functionality
- //
- virtual bool isWindEnabled();
- virtual void enableWind(bool state_b);
-
- // Use these for temporarily muting the audio system.
- // Does not change buffers, initialization, etc. but
- // stops playing new sounds.
- void setMuted(bool muted);
- bool getMuted() const { return mMuted; }
-#ifdef USE_PLUGIN_MEDIA
- LLPluginClassMedia* initializeMedia(const std::string& media_type);
-#endif
- F32 getMasterGain();
- void setMasterGain(F32 gain);
-
- F32 getSecondaryGain(S32 type);
- void setSecondaryGain(S32 type, F32 gain);
-
- F32 getInternetStreamGain();
-
- virtual void setDopplerFactor(F32 factor);
- virtual F32 getDopplerFactor();
- virtual void setRolloffFactor(F32 factor);
- virtual F32 getRolloffFactor();
- virtual void setMaxWindGain(F32 gain);
-
-
- // Methods actually related to setting up and removing sounds
- // Owner ID is the owner of the object making the request
- void triggerSound(const LLUUID &sound_id, const LLUUID& owner_id, const F32 gain,
- const S32 type = LLAudioEngine::AUDIO_TYPE_NONE,
- const LLVector3d &pos_global = LLVector3d::zero);
- void triggerSound(SoundData& soundData);
-
- bool preloadSound(const LLUUID &id);
-
- void addAudioSource(LLAudioSource *asp);
- void cleanupAudioSource(LLAudioSource *asp);
-
- LLAudioSource *findAudioSource(const LLUUID &source_id);
- LLAudioData *getAudioData(const LLUUID &audio_uuid);
-
- // Internet stream implementation manipulation
- LLStreamingAudioInterface *getStreamingAudioImpl();
- void setStreamingAudioImpl(LLStreamingAudioInterface *impl);
- // Internet stream methods - these will call down into the *mStreamingAudioImpl if it exists
- void startInternetStream(const std::string& url);
- void stopInternetStream();
- void pauseInternetStream(S32 pause);
- void updateInternetStream(); // expected to be called often
- LLAudioPlayState isInternetStreamPlaying();
- // use a value from 0.0 to 1.0, inclusive
- void setInternetStreamGain(F32 vol);
- std::string getInternetStreamURL();
-
- // For debugging usage
- virtual LLVector3 getListenerPos();
-
- LLAudioBuffer *getFreeBuffer(); // Get a free buffer, or flush an existing one if you have to.
- LLAudioChannel *getFreeChannel(const F32 priority); // Get a free channel or flush an existing one if your priority is higher
- void cleanupBuffer(LLAudioBuffer *bufferp);
-
- bool hasDecodedFile(const LLUUID &uuid);
- bool hasLocalFile(const LLUUID &uuid);
-
- bool updateBufferForData(LLAudioData *adp, const LLUUID &audio_uuid = LLUUID::null);
-
-
- // Asset callback when we're retrieved a sound from the asset server.
- void startNextTransfer();
- static void assetCallback(const LLUUID &uuid, LLAssetType::EType type, void *user_data, S32 result_code, LLExtStat ext_status);
-
- friend class LLPipeline; // For debugging
-public:
- F32 mMaxWindGain; // Hack. Public to set before fade in?
-
-protected:
- virtual LLAudioBuffer *createBuffer() = 0;
- virtual LLAudioChannel *createChannel() = 0;
-
- virtual bool initWind() = 0;
- virtual void cleanupWind() = 0;
- virtual void setInternalGain(F32 gain) = 0;
-
- void commitDeferredChanges();
-
- virtual void allocateListener() = 0;
-
-
- // listener methods
- virtual void setListenerPos(LLVector3 vec);
- virtual void setListenerVelocity(LLVector3 vec);
- virtual void orientListener(LLVector3 up, LLVector3 at);
- virtual void translateListener(LLVector3 vec);
-
-
- F64 mapWindVecToGain(LLVector3 wind_vec);
- F64 mapWindVecToPitch(LLVector3 wind_vec);
- F64 mapWindVecToPan(LLVector3 wind_vec);
-
-protected:
- LLListener *mListenerp;
-
- bool mMuted;
- void* mUserData;
-
- S32 mLastStatus;
-
- bool mEnableWind;
-
- LLUUID mCurrentTransfer; // Audio file currently being transferred by the system
- LLFrameTimer mCurrentTransferTimer;
-
- // A list of all audio sources that are known to the viewer at this time.
- // This is most likely a superset of the ones that we actually have audio
- // data for, or are playing back.
- typedef std::map<LLUUID, LLAudioSource *> source_map;
- typedef std::map<LLUUID, LLAudioData *> data_map;
-
- source_map mAllSources;
- data_map mAllData;
-
- std::array<LLAudioChannel*, LL_MAX_AUDIO_CHANNELS> mChannels;
-
- // Buffers needs to change into a different data structure, as the number of buffers
- // that we have active should be limited by RAM usage, not count.
- std::array<LLAudioBuffer*, LL_MAX_AUDIO_BUFFERS> mBuffers;
-
- F32 mMasterGain;
- F32 mInternalGain; // Actual gain set; either mMasterGain or 0 when mMuted is true.
- F32 mSecondaryGain[AUDIO_TYPE_COUNT];
-
- F32 mNextWindUpdate;
-
- LLFrameTimer mWindUpdateTimer;
-
-private:
- void setDefaults();
- LLStreamingAudioInterface *mStreamingAudioImpl;
-};
-
-
-
-
-//
-// Standard audio source. Can be derived from for special sources, such as those attached to objects.
-//
-
-
-class LLAudioSource
-{
-public:
- // owner_id is the id of the agent responsible for making this sound
- // play, for example, the owner of the object currently playing it
- LLAudioSource(const LLUUID &id, const LLUUID& owner_id, const F32 gain, const S32 type = LLAudioEngine::AUDIO_TYPE_NONE);
- virtual ~LLAudioSource();
-
- virtual void update(); // Update this audio source
- void updatePriority();
-
- void preload(const LLUUID &audio_id); // Only used for preloading UI sounds, now.
-
- void addAudioData(LLAudioData *adp, bool set_current = true);
-
- void setForcedPriority(const bool ambient) { mForcedPriority = ambient; }
- bool isForcedPriority() const { return mForcedPriority; }
-
- void setLoop(const bool loop) { mLoop = loop; }
- bool isLoop() const { return mLoop; }
-
- void setSyncMaster(const bool master) { mSyncMaster = master; }
- bool isSyncMaster() const { return mSyncMaster; }
-
- void setSyncSlave(const bool slave) { mSyncSlave = slave; }
- bool isSyncSlave() const { return mSyncSlave; }
-
- void setQueueSounds(const bool queue) { mQueueSounds = queue; }
- bool isQueueSounds() const { return mQueueSounds; }
-
- void setPlayedOnce(const bool played_once) { mPlayedOnce = played_once; }
-
- void setType(S32 type) { mType = type; }
- S32 getType() { return mType; }
-
- void setPositionGlobal(const LLVector3d &position_global) { mPositionGlobal = position_global; }
- LLVector3d getPositionGlobal() const { return mPositionGlobal; }
- LLVector3 getVelocity() const { return mVelocity; }
- F32 getPriority() const { return mPriority; }
-
- // Gain should always be clamped between 0 and 1.
- F32 getGain() const { return mGain; }
- virtual void setGain(const F32 gain) { mGain = llclamp(gain, 0.f, 1.f); }
-
- const LLUUID &getID() const { return mID; }
- bool isDone() const;
- bool isMuted() const { return mSourceMuted; }
-
- LLAudioData *getCurrentData();
- LLAudioData *getQueuedData();
- LLAudioBuffer *getCurrentBuffer();
-
- bool setupChannel();
-
- // Stop the audio source, reset audio id even if muted
- void stop();
-
- // Start the audio source playing,
- // takes mute into account to preserve previous id if nessesary
- bool play(const LLUUID &audio_id);
-
- bool hasPendingPreloads() const; // Has preloads that haven't been done yet
-
- friend class LLAudioEngine;
- friend class LLAudioChannel;
-protected:
- void setChannel(LLAudioChannel *channelp);
- LLAudioChannel *getChannel() const { return mChannelp; }
-
-protected:
- LLUUID mID; // The ID of the source is that of the object if it's attached to an object.
- LLUUID mOwnerID; // owner of the object playing the sound
- F32 mPriority;
- F32 mGain;
- bool mSourceMuted;
- bool mForcedPriority; // ignore mute, set high priority, researved for sound preview and UI
- bool mLoop;
- bool mSyncMaster;
- bool mSyncSlave;
- bool mQueueSounds;
- bool mPlayedOnce;
- bool mCorrupted;
- S32 mType;
- LLVector3d mPositionGlobal;
- LLVector3 mVelocity;
-
- //LLAudioSource *mSyncMasterp; // If we're a slave, the source that we're synced to.
- LLAudioChannel *mChannelp; // If we're currently playing back, this is the channel that we're assigned to.
- LLAudioData *mCurrentDatap;
- LLAudioData *mQueuedDatap;
-
- typedef std::map<LLUUID, LLAudioData *> data_map;
- data_map mPreloadMap;
-
- LLFrameTimer mAgeTimer;
-};
-
-
-
-
-//
-// Generic metadata about a particular piece of audio data.
-// The actual data is handled by the derived LLAudioBuffer classes which are
-// derived for each audio engine.
-//
-
-
-class LLAudioData
-{
- public:
- LLAudioData(const LLUUID &uuid);
- bool load();
-
- LLUUID getID() const { return mID; }
- LLAudioBuffer *getBuffer() const { return mBufferp; }
-
- bool hasLocalData() const { return mHasLocalData; }
- bool hasDecodedData() const { return mHasDecodedData; }
- bool hasCompletedDecode() const { return mHasCompletedDecode; }
- bool hasDecodeFailed() const { return mHasDecodeFailed; }
- bool hasWAVLoadFailed() const { return mHasWAVLoadFailed; }
-
- void setHasLocalData(const bool hld) { mHasLocalData = hld; }
- void setHasDecodedData(const bool hdd) { mHasDecodedData = hdd; }
- void setHasCompletedDecode(const bool hcd) { mHasCompletedDecode = hcd; }
- void setHasDecodeFailed(const bool hdf) { mHasDecodeFailed = hdf; }
- void setHasWAVLoadFailed(const bool hwlf) { mHasWAVLoadFailed = hwlf; }
-
- friend class LLAudioEngine; // Severe laziness, bad.
-
- protected:
- LLUUID mID;
- LLAudioBuffer *mBufferp; // If this data is being used by the audio system, a pointer to the buffer will be set here.
- bool mHasLocalData; // Set true if the encoded sound asset file is available locally
- bool mHasDecodedData; // Set true if the decoded sound file is available on disk
- bool mHasCompletedDecode; // Set true when the sound is decoded
- bool mHasDecodeFailed; // Set true if decoding failed, meaning the sound asset is bad
- bool mHasWAVLoadFailed; // Set true if loading the decoded WAV file failed, meaning the sound asset should be decoded instead if
- // possible
-};
-
-
-//
-// Base class for an audio channel, i.e. a channel which is capable of playing back a sound.
-// Management of channels is done generically, methods for actually manipulating the channel
-// are derived for each audio engine.
-//
-
-
-class LLAudioChannel
-{
-public:
- LLAudioChannel();
- virtual ~LLAudioChannel();
-
- virtual void setSource(LLAudioSource *sourcep);
- LLAudioSource *getSource() const { return mCurrentSourcep; }
-
- void setSecondaryGain(F32 gain) { mSecondaryGain = gain; }
- F32 getSecondaryGain() { return mSecondaryGain; }
-
- friend class LLAudioEngine;
- friend class LLAudioSource;
-protected:
- virtual void play() = 0;
- virtual void playSynced(LLAudioChannel *channelp) = 0;
- virtual void cleanup() = 0;
- virtual bool isPlaying() = 0;
- void setWaiting(const bool waiting) { mWaiting = waiting; }
- bool isWaiting() const { return mWaiting; }
-
- virtual bool updateBuffer(); // Check to see if the buffer associated with the source changed, and update if necessary.
- virtual void update3DPosition() = 0;
- virtual void updateLoop() = 0; // Update your loop/completion status, for use by queueing/syncing.
-protected:
- LLAudioSource *mCurrentSourcep;
- LLAudioBuffer *mCurrentBufferp;
- bool mLoopedThisFrame;
- bool mWaiting; // Waiting for sync.
- F32 mSecondaryGain;
-};
-
-
-
-
-// Basically an interface class to the engine-specific implementation
-// of audio data that's ready for playback.
-// Will likely get more complex as we decide to do stuff like real streaming audio.
-
-
-class LLAudioBuffer
-{
-public:
- virtual ~LLAudioBuffer() {};
- virtual bool loadWAV(const std::string& filename) = 0;
- virtual U32 getLength() = 0;
-
- friend class LLAudioEngine;
- friend class LLAudioChannel;
- friend class LLAudioData;
-protected:
- bool mInUse;
- LLAudioData *mAudioDatap;
- LLFrameTimer mLastUseTimer;
-};
-
-struct SoundData
-{
- LLUUID audio_uuid;
- LLUUID owner_id;
- F32 gain;
- S32 type;
- LLVector3d pos_global;
-
- SoundData(const LLUUID &audio_uuid,
- const LLUUID& owner_id,
- const F32 gain,
- const S32 type = LLAudioEngine::AUDIO_TYPE_NONE,
- const LLVector3d &pos_global = LLVector3d::zero) :
- audio_uuid(audio_uuid),
- owner_id(owner_id),
- gain(gain),
- type(type),
- pos_global(pos_global)
- {
- }
-};
-
-
-extern LLAudioEngine* gAudiop;
-
-#endif
+/**
+ * @file audioengine.h
+ * @brief Definition of LLAudioEngine base class abstracting the audio support
+ *
+ * $LicenseInfo:firstyear=2000&license=viewerlgpl$
+ * Second Life Viewer Source Code
+ * Copyright (C) 2010, Linden Research, Inc.
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation;
+ * version 2.1 of the License only.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with this library; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ *
+ * Linden Research, Inc., 945 Battery Street, San Francisco, CA 94111 USA
+ * $/LicenseInfo$
+ */
+
+
+#ifndef LL_AUDIOENGINE_H
+#define LL_AUDIOENGINE_H
+
+#include <list>
+#include <map>
+#include <array>
+
+#include "v3math.h"
+#include "v3dmath.h"
+#include "lltimer.h"
+#include "lluuid.h"
+#include "llframetimer.h"
+#include "llassettype.h"
+#include "llextendedstatus.h"
+
+#include "lllistener.h"
+
+const F32 LL_WIND_UPDATE_INTERVAL = 0.1f;
+const F32 LL_WIND_UNDERWATER_CENTER_FREQ = 20.f;
+
+const F32 ATTACHED_OBJECT_TIMEOUT = 5.0f;
+const F32 DEFAULT_MIN_DISTANCE = 2.0f;
+
+#define LL_MAX_AUDIO_CHANNELS 30
+#define LL_MAX_AUDIO_BUFFERS 40 // Some extra for preloading, maybe?
+
+class LLAudioSource;
+class LLAudioData;
+class LLAudioChannel;
+class LLAudioChannelOpenAL;
+class LLAudioBuffer;
+class LLStreamingAudioInterface;
+struct SoundData;
+
+//
+// LLAudioEngine definition
+//
+class LLAudioEngine
+{
+ friend class LLAudioChannelOpenAL; // bleh. channel needs some listener methods.
+
+public:
+ enum LLAudioType
+ {
+ AUDIO_TYPE_NONE = 0,
+ AUDIO_TYPE_SFX = 1,
+ AUDIO_TYPE_UI = 2,
+ AUDIO_TYPE_AMBIENT = 3,
+ AUDIO_TYPE_COUNT = 4 // last
+ };
+
+ enum LLAudioPlayState
+ {
+ // isInternetStreamPlaying() returns an *S32*, with
+ // 0 = stopped, 1 = playing, 2 = paused.
+ AUDIO_STOPPED = 0,
+ AUDIO_PLAYING = 1,
+ AUDIO_PAUSED = 2
+ };
+
+ LLAudioEngine();
+ virtual ~LLAudioEngine();
+
+ // initialization/startup/shutdown
+ virtual bool init(void *userdata, const std::string &app_title);
+ virtual std::string getDriverName(bool verbose) = 0;
+ virtual LLStreamingAudioInterface *createDefaultStreamingAudioImpl() const = 0;
+ virtual void shutdown();
+
+ // Used by the mechanics of the engine
+ //virtual void processQueue(const LLUUID &sound_guid);
+ virtual void setListener(LLVector3 pos,LLVector3 vel,LLVector3 up,LLVector3 at);
+ virtual void updateWind(LLVector3 direction, F32 camera_height_above_water) = 0;
+ virtual void idle();
+ virtual void updateChannels();
+
+ //
+ // "End user" functionality
+ //
+ virtual bool isWindEnabled();
+ virtual void enableWind(bool state_b);
+
+ // Use these for temporarily muting the audio system.
+ // Does not change buffers, initialization, etc. but
+ // stops playing new sounds.
+ void setMuted(bool muted);
+ bool getMuted() const { return mMuted; }
+#ifdef USE_PLUGIN_MEDIA
+ LLPluginClassMedia* initializeMedia(const std::string& media_type);
+#endif
+ F32 getMasterGain();
+ void setMasterGain(F32 gain);
+
+ F32 getSecondaryGain(S32 type);
+ void setSecondaryGain(S32 type, F32 gain);
+
+ F32 getInternetStreamGain();
+
+ virtual void setDopplerFactor(F32 factor);
+ virtual F32 getDopplerFactor();
+ virtual void setRolloffFactor(F32 factor);
+ virtual F32 getRolloffFactor();
+ virtual void setMaxWindGain(F32 gain);
+
+
+ // Methods actually related to setting up and removing sounds
+ // Owner ID is the owner of the object making the request
+ void triggerSound(const LLUUID &sound_id, const LLUUID& owner_id, const F32 gain,
+ const S32 type = LLAudioEngine::AUDIO_TYPE_NONE,
+ const LLVector3d &pos_global = LLVector3d::zero);
+ void triggerSound(SoundData& soundData);
+
+ bool preloadSound(const LLUUID &id);
+
+ void addAudioSource(LLAudioSource *asp);
+ void cleanupAudioSource(LLAudioSource *asp);
+
+ LLAudioSource *findAudioSource(const LLUUID &source_id);
+ LLAudioData *getAudioData(const LLUUID &audio_uuid);
+
+ // Internet stream implementation manipulation
+ LLStreamingAudioInterface *getStreamingAudioImpl();
+ void setStreamingAudioImpl(LLStreamingAudioInterface *impl);
+ // Internet stream methods - these will call down into the *mStreamingAudioImpl if it exists
+ void startInternetStream(const std::string& url);
+ void stopInternetStream();
+ void pauseInternetStream(S32 pause);
+ void updateInternetStream(); // expected to be called often
+ LLAudioPlayState isInternetStreamPlaying();
+ // use a value from 0.0 to 1.0, inclusive
+ void setInternetStreamGain(F32 vol);
+ std::string getInternetStreamURL();
+
+ // For debugging usage
+ virtual LLVector3 getListenerPos();
+
+ LLAudioBuffer *getFreeBuffer(); // Get a free buffer, or flush an existing one if you have to.
+ LLAudioChannel *getFreeChannel(const F32 priority); // Get a free channel or flush an existing one if your priority is higher
+ void cleanupBuffer(LLAudioBuffer *bufferp);
+
+ bool hasDecodedFile(const LLUUID &uuid);
+ bool hasLocalFile(const LLUUID &uuid);
+
+ bool updateBufferForData(LLAudioData *adp, const LLUUID &audio_uuid = LLUUID::null);
+
+
+ // Asset callback when we're retrieved a sound from the asset server.
+ void startNextTransfer();
+ static void assetCallback(const LLUUID &uuid, LLAssetType::EType type, void *user_data, S32 result_code, LLExtStat ext_status);
+
+ friend class LLPipeline; // For debugging
+public:
+ F32 mMaxWindGain; // Hack. Public to set before fade in?
+
+protected:
+ virtual LLAudioBuffer *createBuffer() = 0;
+ virtual LLAudioChannel *createChannel() = 0;
+
+ virtual bool initWind() = 0;
+ virtual void cleanupWind() = 0;
+ virtual void setInternalGain(F32 gain) = 0;
+
+ void commitDeferredChanges();
+
+ virtual void allocateListener() = 0;
+
+
+ // listener methods
+ virtual void setListenerPos(LLVector3 vec);
+ virtual void setListenerVelocity(LLVector3 vec);
+ virtual void orientListener(LLVector3 up, LLVector3 at);
+ virtual void translateListener(LLVector3 vec);
+
+
+ F64 mapWindVecToGain(LLVector3 wind_vec);
+ F64 mapWindVecToPitch(LLVector3 wind_vec);
+ F64 mapWindVecToPan(LLVector3 wind_vec);
+
+protected:
+ LLListener *mListenerp;
+
+ bool mMuted;
+ void* mUserData;
+
+ S32 mLastStatus;
+
+ bool mEnableWind;
+
+ LLUUID mCurrentTransfer; // Audio file currently being transferred by the system
+ LLFrameTimer mCurrentTransferTimer;
+
+ // A list of all audio sources that are known to the viewer at this time.
+ // This is most likely a superset of the ones that we actually have audio
+ // data for, or are playing back.
+ typedef std::map<LLUUID, LLAudioSource *> source_map;
+ typedef std::map<LLUUID, LLAudioData *> data_map;
+
+ source_map mAllSources;
+ data_map mAllData;
+
+ std::array<LLAudioChannel*, LL_MAX_AUDIO_CHANNELS> mChannels;
+
+ // Buffers needs to change into a different data structure, as the number of buffers
+ // that we have active should be limited by RAM usage, not count.
+ std::array<LLAudioBuffer*, LL_MAX_AUDIO_BUFFERS> mBuffers;
+
+ F32 mMasterGain;
+ F32 mInternalGain; // Actual gain set; either mMasterGain or 0 when mMuted is true.
+ F32 mSecondaryGain[AUDIO_TYPE_COUNT];
+
+ F32 mNextWindUpdate;
+
+ LLFrameTimer mWindUpdateTimer;
+
+private:
+ void setDefaults();
+ LLStreamingAudioInterface *mStreamingAudioImpl;
+};
+
+
+
+
+//
+// Standard audio source. Can be derived from for special sources, such as those attached to objects.
+//
+
+
+class LLAudioSource
+{
+public:
+ // owner_id is the id of the agent responsible for making this sound
+ // play, for example, the owner of the object currently playing it
+ LLAudioSource(const LLUUID &id, const LLUUID& owner_id, const F32 gain, const S32 type = LLAudioEngine::AUDIO_TYPE_NONE);
+ virtual ~LLAudioSource();
+
+ virtual void update(); // Update this audio source
+ void updatePriority();
+
+ void preload(const LLUUID &audio_id); // Only used for preloading UI sounds, now.
+
+ void addAudioData(LLAudioData *adp, bool set_current = true);
+
+ void setForcedPriority(const bool ambient) { mForcedPriority = ambient; }
+ bool isForcedPriority() const { return mForcedPriority; }
+
+ void setLoop(const bool loop) { mLoop = loop; }
+ bool isLoop() const { return mLoop; }
+
+ void setSyncMaster(const bool master) { mSyncMaster = master; }
+ bool isSyncMaster() const { return mSyncMaster; }
+
+ void setSyncSlave(const bool slave) { mSyncSlave = slave; }
+ bool isSyncSlave() const { return mSyncSlave; }
+
+ void setQueueSounds(const bool queue) { mQueueSounds = queue; }
+ bool isQueueSounds() const { return mQueueSounds; }
+
+ void setPlayedOnce(const bool played_once) { mPlayedOnce = played_once; }
+
+ void setType(S32 type) { mType = type; }
+ S32 getType() { return mType; }
+
+ void setPositionGlobal(const LLVector3d &position_global) { mPositionGlobal = position_global; }
+ LLVector3d getPositionGlobal() const { return mPositionGlobal; }
+ LLVector3 getVelocity() const { return mVelocity; }
+ F32 getPriority() const { return mPriority; }
+
+ // Gain should always be clamped between 0 and 1.
+ F32 getGain() const { return mGain; }
+ virtual void setGain(const F32 gain) { mGain = llclamp(gain, 0.f, 1.f); }
+
+ const LLUUID &getID() const { return mID; }
+ bool isDone() const;
+ bool isMuted() const { return mSourceMuted; }
+
+ LLAudioData *getCurrentData();
+ LLAudioData *getQueuedData();
+ LLAudioBuffer *getCurrentBuffer();
+
+ bool setupChannel();
+
+ // Stop the audio source, reset audio id even if muted
+ void stop();
+
+ // Start the audio source playing,
+ // takes mute into account to preserve previous id if nessesary
+ bool play(const LLUUID &audio_id);
+
+ bool hasPendingPreloads() const; // Has preloads that haven't been done yet
+
+ friend class LLAudioEngine;
+ friend class LLAudioChannel;
+protected:
+ void setChannel(LLAudioChannel *channelp);
+ LLAudioChannel *getChannel() const { return mChannelp; }
+
+protected:
+ LLUUID mID; // The ID of the source is that of the object if it's attached to an object.
+ LLUUID mOwnerID; // owner of the object playing the sound
+ F32 mPriority;
+ F32 mGain;
+ bool mSourceMuted;
+ bool mForcedPriority; // ignore mute, set high priority, researved for sound preview and UI
+ bool mLoop;
+ bool mSyncMaster;
+ bool mSyncSlave;
+ bool mQueueSounds;
+ bool mPlayedOnce;
+ bool mCorrupted;
+ S32 mType;
+ LLVector3d mPositionGlobal;
+ LLVector3 mVelocity;
+
+ //LLAudioSource *mSyncMasterp; // If we're a slave, the source that we're synced to.
+ LLAudioChannel *mChannelp; // If we're currently playing back, this is the channel that we're assigned to.
+ LLAudioData *mCurrentDatap;
+ LLAudioData *mQueuedDatap;
+
+ typedef std::map<LLUUID, LLAudioData *> data_map;
+ data_map mPreloadMap;
+
+ LLFrameTimer mAgeTimer;
+};
+
+
+
+
+//
+// Generic metadata about a particular piece of audio data.
+// The actual data is handled by the derived LLAudioBuffer classes which are
+// derived for each audio engine.
+//
+
+
+class LLAudioData
+{
+ public:
+ LLAudioData(const LLUUID &uuid);
+ bool load();
+
+ LLUUID getID() const { return mID; }
+ LLAudioBuffer *getBuffer() const { return mBufferp; }
+
+ bool hasLocalData() const { return mHasLocalData; }
+ bool hasDecodedData() const { return mHasDecodedData; }
+ bool hasCompletedDecode() const { return mHasCompletedDecode; }
+ bool hasDecodeFailed() const { return mHasDecodeFailed; }
+ bool hasWAVLoadFailed() const { return mHasWAVLoadFailed; }
+
+ void setHasLocalData(const bool hld) { mHasLocalData = hld; }
+ void setHasDecodedData(const bool hdd) { mHasDecodedData = hdd; }
+ void setHasCompletedDecode(const bool hcd) { mHasCompletedDecode = hcd; }
+ void setHasDecodeFailed(const bool hdf) { mHasDecodeFailed = hdf; }
+ void setHasWAVLoadFailed(const bool hwlf) { mHasWAVLoadFailed = hwlf; }
+
+ friend class LLAudioEngine; // Severe laziness, bad.
+
+ protected:
+ LLUUID mID;
+ LLAudioBuffer *mBufferp; // If this data is being used by the audio system, a pointer to the buffer will be set here.
+ bool mHasLocalData; // Set true if the encoded sound asset file is available locally
+ bool mHasDecodedData; // Set true if the decoded sound file is available on disk
+ bool mHasCompletedDecode; // Set true when the sound is decoded
+ bool mHasDecodeFailed; // Set true if decoding failed, meaning the sound asset is bad
+ bool mHasWAVLoadFailed; // Set true if loading the decoded WAV file failed, meaning the sound asset should be decoded instead if
+ // possible
+};
+
+
+//
+// Base class for an audio channel, i.e. a channel which is capable of playing back a sound.
+// Management of channels is done generically, methods for actually manipulating the channel
+// are derived for each audio engine.
+//
+
+
+class LLAudioChannel
+{
+public:
+ LLAudioChannel();
+ virtual ~LLAudioChannel();
+
+ virtual void setSource(LLAudioSource *sourcep);
+ LLAudioSource *getSource() const { return mCurrentSourcep; }
+
+ void setSecondaryGain(F32 gain) { mSecondaryGain = gain; }
+ F32 getSecondaryGain() { return mSecondaryGain; }
+
+ friend class LLAudioEngine;
+ friend class LLAudioSource;
+protected:
+ virtual void play() = 0;
+ virtual void playSynced(LLAudioChannel *channelp) = 0;
+ virtual void cleanup() = 0;
+ virtual bool isPlaying() = 0;
+ void setWaiting(const bool waiting) { mWaiting = waiting; }
+ bool isWaiting() const { return mWaiting; }
+
+ virtual bool updateBuffer(); // Check to see if the buffer associated with the source changed, and update if necessary.
+ virtual void update3DPosition() = 0;
+ virtual void updateLoop() = 0; // Update your loop/completion status, for use by queueing/syncing.
+protected:
+ LLAudioSource *mCurrentSourcep;
+ LLAudioBuffer *mCurrentBufferp;
+ bool mLoopedThisFrame;
+ bool mWaiting; // Waiting for sync.
+ F32 mSecondaryGain;
+};
+
+
+
+
+// Basically an interface class to the engine-specific implementation
+// of audio data that's ready for playback.
+// Will likely get more complex as we decide to do stuff like real streaming audio.
+
+
+class LLAudioBuffer
+{
+public:
+ virtual ~LLAudioBuffer() {};
+ virtual bool loadWAV(const std::string& filename) = 0;
+ virtual U32 getLength() = 0;
+
+ friend class LLAudioEngine;
+ friend class LLAudioChannel;
+ friend class LLAudioData;
+protected:
+ bool mInUse;
+ LLAudioData *mAudioDatap;
+ LLFrameTimer mLastUseTimer;
+};
+
+struct SoundData
+{
+ LLUUID audio_uuid;
+ LLUUID owner_id;
+ F32 gain;
+ S32 type;
+ LLVector3d pos_global;
+
+ SoundData(const LLUUID &audio_uuid,
+ const LLUUID& owner_id,
+ const F32 gain,
+ const S32 type = LLAudioEngine::AUDIO_TYPE_NONE,
+ const LLVector3d &pos_global = LLVector3d::zero) :
+ audio_uuid(audio_uuid),
+ owner_id(owner_id),
+ gain(gain),
+ type(type),
+ pos_global(pos_global)
+ {
+ }
+};
+
+
+extern LLAudioEngine* gAudiop;
+
+#endif
diff --git a/indra/llaudio/llaudioengine_openal.cpp b/indra/llaudio/llaudioengine_openal.cpp
index 9e62e9bfec..18d682b554 100644
--- a/indra/llaudio/llaudioengine_openal.cpp
+++ b/indra/llaudio/llaudioengine_openal.cpp
@@ -1,560 +1,560 @@
-/**
- * @file audioengine_openal.cpp
- * @brief implementation of audio engine using OpenAL
- * support as a OpenAL 3D implementation
- *
- * $LicenseInfo:firstyear=2002&license=viewerlgpl$
- * Second Life Viewer Source Code
- * Copyright (C) 2010, Linden Research, Inc.
- *
- * This library is free software; you can redistribute it and/or
- * modify it under the terms of the GNU Lesser General Public
- * License as published by the Free Software Foundation;
- * version 2.1 of the License only.
- *
- * This library is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
- * Lesser General Public License for more details.
- *
- * You should have received a copy of the GNU Lesser General Public
- * License along with this library; if not, write to the Free Software
- * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
- *
- * Linden Research, Inc., 945 Battery Street, San Francisco, CA 94111 USA
- * $/LicenseInfo$
- */
-
-#include "linden_common.h"
-#include "lldir.h"
-
-#include "llaudioengine_openal.h"
-#include "lllistener_openal.h"
-
-
-const float LLAudioEngine_OpenAL::WIND_BUFFER_SIZE_SEC = 0.05f;
-
-LLAudioEngine_OpenAL::LLAudioEngine_OpenAL()
- :
- mWindGen(NULL),
- mWindBuf(NULL),
- mWindBufFreq(0),
- mWindBufSamples(0),
- mWindBufBytes(0),
- mWindSource(AL_NONE),
- mNumEmptyWindALBuffers(MAX_NUM_WIND_BUFFERS)
-{
-}
-
-// virtual
-LLAudioEngine_OpenAL::~LLAudioEngine_OpenAL()
-{
-}
-
-// virtual
-bool LLAudioEngine_OpenAL::init(void* userdata, const std::string &app_title)
-{
- mWindGen = NULL;
- LLAudioEngine::init(userdata, app_title);
-
- if(!alutInit(NULL, NULL))
- {
- LL_WARNS() << "LLAudioEngine_OpenAL::init() ALUT initialization failed: " << alutGetErrorString (alutGetError ()) << LL_ENDL;
- return false;
- }
-
- LL_INFOS() << "LLAudioEngine_OpenAL::init() OpenAL successfully initialized" << LL_ENDL;
-
- LL_INFOS() << "OpenAL version: "
- << ll_safe_string(alGetString(AL_VERSION)) << LL_ENDL;
- LL_INFOS() << "OpenAL vendor: "
- << ll_safe_string(alGetString(AL_VENDOR)) << LL_ENDL;
- LL_INFOS() << "OpenAL renderer: "
- << ll_safe_string(alGetString(AL_RENDERER)) << LL_ENDL;
-
- ALint major = alutGetMajorVersion ();
- ALint minor = alutGetMinorVersion ();
- LL_INFOS() << "ALUT version: " << major << "." << minor << LL_ENDL;
-
- ALCdevice *device = alcGetContextsDevice(alcGetCurrentContext());
-
- alcGetIntegerv(device, ALC_MAJOR_VERSION, 1, &major);
- alcGetIntegerv(device, ALC_MAJOR_VERSION, 1, &minor);
- LL_INFOS() << "ALC version: " << major << "." << minor << LL_ENDL;
-
- LL_INFOS() << "ALC default device: "
- << ll_safe_string(alcGetString(device,
- ALC_DEFAULT_DEVICE_SPECIFIER))
- << LL_ENDL;
-
- return true;
-}
-
-// virtual
-std::string LLAudioEngine_OpenAL::getDriverName(bool verbose)
-{
- ALCdevice *device = alcGetContextsDevice(alcGetCurrentContext());
- std::ostringstream version;
-
- version <<
- "OpenAL";
-
- if (verbose)
- {
- version <<
- ", version " <<
- ll_safe_string(alGetString(AL_VERSION)) <<
- " / " <<
- ll_safe_string(alGetString(AL_VENDOR)) <<
- " / " <<
- ll_safe_string(alGetString(AL_RENDERER));
-
- if (device)
- version <<
- ": " <<
- ll_safe_string(alcGetString(device,
- ALC_DEFAULT_DEVICE_SPECIFIER));
- }
-
- return version.str();
-}
-
-// virtual
-void LLAudioEngine_OpenAL::allocateListener()
-{
- mListenerp = (LLListener *) new LLListener_OpenAL();
- if(!mListenerp)
- {
- LL_WARNS() << "LLAudioEngine_OpenAL::allocateListener() Listener creation failed" << LL_ENDL;
- }
-}
-
-// virtual
-void LLAudioEngine_OpenAL::shutdown()
-{
- LL_INFOS() << "About to LLAudioEngine::shutdown()" << LL_ENDL;
- LLAudioEngine::shutdown();
-
- // If a subsequent error occurs while there is still an error recorded
- // internally, the second error will simply be ignored.
- // Clear previous error to make sure we will captuare a valid failure reason
- ALenum error = alutGetError();
- if (error != ALUT_ERROR_NO_ERROR)
- {
- LL_WARNS() << "Uncleared error state prior to shutdown: "
- << alutGetErrorString(error) << LL_ENDL;
- }
-
- LL_INFOS() << "About to alutExit()" << LL_ENDL;
- if(!alutExit())
- {
- LL_WARNS() << "LLAudioEngine_OpenAL::shutdown() ALUT shutdown failed: " << alutGetErrorString (alutGetError ()) << LL_ENDL;
- }
-
- LL_INFOS() << "LLAudioEngine_OpenAL::shutdown() OpenAL successfully shut down" << LL_ENDL;
-
- delete mListenerp;
- mListenerp = NULL;
-}
-
-LLAudioBuffer *LLAudioEngine_OpenAL::createBuffer()
-{
- return new LLAudioBufferOpenAL();
-}
-
-LLAudioChannel *LLAudioEngine_OpenAL::createChannel()
-{
- return new LLAudioChannelOpenAL();
-}
-
-void LLAudioEngine_OpenAL::setInternalGain(F32 gain)
-{
- //LL_INFOS() << "LLAudioEngine_OpenAL::setInternalGain() Gain: " << gain << LL_ENDL;
- alListenerf(AL_GAIN, gain);
-}
-
-LLAudioChannelOpenAL::LLAudioChannelOpenAL()
- :
- mALSource(AL_NONE),
- mLastSamplePos(0)
-{
- alGenSources(1, &mALSource);
-}
-
-LLAudioChannelOpenAL::~LLAudioChannelOpenAL()
-{
- cleanup();
- alDeleteSources(1, &mALSource);
-}
-
-void LLAudioChannelOpenAL::cleanup()
-{
- alSourceStop(mALSource);
- mCurrentBufferp = NULL;
-}
-
-void LLAudioChannelOpenAL::play()
-{
- if (mALSource == AL_NONE)
- {
- LL_WARNS() << "Playing without a mALSource, aborting" << LL_ENDL;
- return;
- }
-
- if(!isPlaying())
- {
- alSourcePlay(mALSource);
- getSource()->setPlayedOnce(true);
- }
-}
-
-void LLAudioChannelOpenAL::playSynced(LLAudioChannel *channelp)
-{
- if (channelp)
- {
- LLAudioChannelOpenAL *masterchannelp =
- (LLAudioChannelOpenAL*)channelp;
- if (mALSource != AL_NONE &&
- masterchannelp->mALSource != AL_NONE)
- {
- // we have channels allocated to master and slave
- ALfloat master_offset;
- alGetSourcef(masterchannelp->mALSource, AL_SEC_OFFSET,
- &master_offset);
-
- LL_INFOS() << "Syncing with master at " << master_offset
- << "sec" << LL_ENDL;
- // *TODO: detect when this fails, maybe use AL_SAMPLE_
- alSourcef(mALSource, AL_SEC_OFFSET, master_offset);
- }
- }
- play();
-}
-
-bool LLAudioChannelOpenAL::isPlaying()
-{
- if (mALSource != AL_NONE)
- {
- ALint state;
- alGetSourcei(mALSource, AL_SOURCE_STATE, &state);
- if(state == AL_PLAYING)
- {
- return true;
- }
- }
-
- return false;
-}
-
-bool LLAudioChannelOpenAL::updateBuffer()
-{
- if (!mCurrentSourcep)
- {
- // This channel isn't associated with any source, nothing
- // to be updated
- return false;
- }
-
- if (LLAudioChannel::updateBuffer())
- {
- // Base class update returned true, which means that we need to actually
- // set up the source for a different buffer.
- LLAudioBufferOpenAL *bufferp = (LLAudioBufferOpenAL *)mCurrentSourcep->getCurrentBuffer();
- ALuint buffer = bufferp->getBuffer();
- alSourcei(mALSource, AL_BUFFER, buffer);
- mLastSamplePos = 0;
- }
-
- if (mCurrentSourcep)
- {
- alSourcef(mALSource, AL_GAIN,
- mCurrentSourcep->getGain() * getSecondaryGain());
- alSourcei(mALSource, AL_LOOPING,
- mCurrentSourcep->isLoop() ? AL_TRUE : AL_FALSE);
- alSourcef(mALSource, AL_ROLLOFF_FACTOR,
- gAudiop->mListenerp->getRolloffFactor());
- }
-
- return true;
-}
-
-
-void LLAudioChannelOpenAL::updateLoop()
-{
- if (mALSource == AL_NONE)
- {
- return;
- }
-
- // Hack: We keep track of whether we looped or not by seeing when the
- // sample position looks like it's going backwards. Not reliable; may
- // yield false negatives.
- //
- ALint cur_pos;
- alGetSourcei(mALSource, AL_SAMPLE_OFFSET, &cur_pos);
- if (cur_pos < mLastSamplePos)
- {
- mLoopedThisFrame = true;
- }
- mLastSamplePos = cur_pos;
-}
-
-
-void LLAudioChannelOpenAL::update3DPosition()
-{
- if(!mCurrentSourcep)
- {
- return;
- }
- if (mCurrentSourcep->isForcedPriority())
- {
- alSource3f(mALSource, AL_POSITION, 0.0, 0.0, 0.0);
- alSource3f(mALSource, AL_VELOCITY, 0.0, 0.0, 0.0);
- alSourcei (mALSource, AL_SOURCE_RELATIVE, AL_TRUE);
- } else {
- LLVector3 float_pos;
- float_pos.setVec(mCurrentSourcep->getPositionGlobal());
- alSourcefv(mALSource, AL_POSITION, float_pos.mV);
- alSourcefv(mALSource, AL_VELOCITY, mCurrentSourcep->getVelocity().mV);
- alSourcei (mALSource, AL_SOURCE_RELATIVE, AL_FALSE);
- }
-
- alSourcef(mALSource, AL_GAIN, mCurrentSourcep->getGain() * getSecondaryGain());
-}
-
-LLAudioBufferOpenAL::LLAudioBufferOpenAL()
-{
- mALBuffer = AL_NONE;
-}
-
-LLAudioBufferOpenAL::~LLAudioBufferOpenAL()
-{
- cleanup();
-}
-
-void LLAudioBufferOpenAL::cleanup()
-{
- if(mALBuffer != AL_NONE)
- {
- alDeleteBuffers(1, &mALBuffer);
- mALBuffer = AL_NONE;
- }
-}
-
-bool LLAudioBufferOpenAL::loadWAV(const std::string& filename)
-{
- cleanup();
- mALBuffer = alutCreateBufferFromFile(filename.c_str());
- if(mALBuffer == AL_NONE)
- {
- ALenum error = alutGetError();
- if (gDirUtilp->fileExists(filename))
- {
- LL_WARNS() <<
- "LLAudioBufferOpenAL::loadWAV() Error loading "
- << filename
- << " " << alutGetErrorString(error) << LL_ENDL;
- }
- else
- {
- // It's common for the file to not actually exist.
- LL_DEBUGS() <<
- "LLAudioBufferOpenAL::loadWAV() Error loading "
- << filename
- << " " << alutGetErrorString(error) << LL_ENDL;
- }
- return false;
- }
-
- return true;
-}
-
-U32 LLAudioBufferOpenAL::getLength()
-{
- if(mALBuffer == AL_NONE)
- {
- return 0;
- }
- ALint length;
- alGetBufferi(mALBuffer, AL_SIZE, &length);
- return length / 2; // convert size in bytes to size in (16-bit) samples
-}
-
-// ------------
-
-bool LLAudioEngine_OpenAL::initWind()
-{
- ALenum error;
- LL_INFOS() << "LLAudioEngine_OpenAL::initWind() start" << LL_ENDL;
-
- mNumEmptyWindALBuffers = MAX_NUM_WIND_BUFFERS;
-
- alGetError(); /* clear error */
-
- alGenSources(1,&mWindSource);
-
- if((error=alGetError()) != AL_NO_ERROR)
- {
- LL_WARNS() << "LLAudioEngine_OpenAL::initWind() Error creating wind sources: "<<error<<LL_ENDL;
- }
-
- mWindGen = new LLWindGen<WIND_SAMPLE_T>;
-
- mWindBufFreq = mWindGen->getInputSamplingRate();
- mWindBufSamples = llceil(mWindBufFreq * WIND_BUFFER_SIZE_SEC);
- mWindBufBytes = mWindBufSamples * 2 /*stereo*/ * sizeof(WIND_SAMPLE_T);
-
- mWindBuf = new WIND_SAMPLE_T [mWindBufSamples * 2 /*stereo*/];
-
- if(mWindBuf==NULL)
- {
- LL_ERRS() << "LLAudioEngine_OpenAL::initWind() Error creating wind memory buffer" << LL_ENDL;
- return false;
- }
-
- LL_INFOS() << "LLAudioEngine_OpenAL::initWind() done" << LL_ENDL;
-
- return true;
-}
-
-void LLAudioEngine_OpenAL::cleanupWind()
-{
- LL_INFOS() << "LLAudioEngine_OpenAL::cleanupWind()" << LL_ENDL;
-
- if (mWindSource != AL_NONE)
- {
- // detach and delete all outstanding buffers on the wind source
- alSourceStop(mWindSource);
- ALint processed;
- alGetSourcei(mWindSource, AL_BUFFERS_PROCESSED, &processed);
- while (processed--)
- {
- ALuint buffer = AL_NONE;
- alSourceUnqueueBuffers(mWindSource, 1, &buffer);
- alDeleteBuffers(1, &buffer);
- }
-
- // delete the wind source itself
- alDeleteSources(1, &mWindSource);
-
- mWindSource = AL_NONE;
- }
-
- delete[] mWindBuf;
- mWindBuf = NULL;
-
- delete mWindGen;
- mWindGen = NULL;
-}
-
-void LLAudioEngine_OpenAL::updateWind(LLVector3 wind_vec, F32 camera_altitude)
-{
- LLVector3 wind_pos;
- F64 pitch;
- F64 center_freq;
- ALenum error;
-
- if (!mEnableWind)
- return;
-
- if(!mWindBuf)
- return;
-
- if (mWindUpdateTimer.checkExpirationAndReset(LL_WIND_UPDATE_INTERVAL))
- {
-
- // wind comes in as Linden coordinate (+X = forward, +Y = left, +Z = up)
- // need to convert this to the conventional orientation DS3D and OpenAL use
- // where +X = right, +Y = up, +Z = backwards
-
- wind_vec.setVec(-wind_vec.mV[1], wind_vec.mV[2], -wind_vec.mV[0]);
-
- pitch = 1.0 + mapWindVecToPitch(wind_vec);
- center_freq = 80.0 * pow(pitch,2.5*(mapWindVecToGain(wind_vec)+1.0));
-
- mWindGen->mTargetFreq = (F32)center_freq;
- mWindGen->mTargetGain = (F32)mapWindVecToGain(wind_vec) * mMaxWindGain;
- mWindGen->mTargetPanGainR = (F32)mapWindVecToPan(wind_vec);
-
- alSourcei(mWindSource, AL_LOOPING, AL_FALSE);
- alSource3f(mWindSource, AL_POSITION, 0.0, 0.0, 0.0);
- alSource3f(mWindSource, AL_VELOCITY, 0.0, 0.0, 0.0);
- alSourcef(mWindSource, AL_ROLLOFF_FACTOR, 0.0);
- alSourcei(mWindSource, AL_SOURCE_RELATIVE, AL_TRUE);
- }
-
- // ok lets make a wind buffer now
-
- ALint processed, queued, unprocessed;
- alGetSourcei(mWindSource, AL_BUFFERS_PROCESSED, &processed);
- alGetSourcei(mWindSource, AL_BUFFERS_QUEUED, &queued);
- unprocessed = queued - processed;
-
- // ensure that there are always at least 3x as many filled buffers
- // queued as we managed to empty since last time.
- mNumEmptyWindALBuffers = llmin(mNumEmptyWindALBuffers + processed * 3 - unprocessed, MAX_NUM_WIND_BUFFERS-unprocessed);
- mNumEmptyWindALBuffers = llmax(mNumEmptyWindALBuffers, 0);
-
- //LL_INFOS() << "mNumEmptyWindALBuffers: " << mNumEmptyWindALBuffers <<" (" << unprocessed << ":" << processed << ")" << LL_ENDL;
-
- while(processed--) // unqueue old buffers
- {
- ALuint buffer;
- ALenum error;
- alGetError(); /* clear error */
- alSourceUnqueueBuffers(mWindSource, 1, &buffer);
- error = alGetError();
- if(error != AL_NO_ERROR)
- {
- LL_WARNS() << "LLAudioEngine_OpenAL::updateWind() error swapping (unqueuing) buffers" << LL_ENDL;
- }
- else
- {
- alDeleteBuffers(1, &buffer);
- }
- }
-
- unprocessed += mNumEmptyWindALBuffers;
- while (mNumEmptyWindALBuffers > 0) // fill+queue new buffers
- {
- ALuint buffer;
- alGetError(); /* clear error */
- alGenBuffers(1,&buffer);
- if((error=alGetError()) != AL_NO_ERROR)
- {
- LL_WARNS() << "LLAudioEngine_OpenAL::updateWind() Error creating wind buffer: " << error << LL_ENDL;
- break;
- }
-
- alBufferData(buffer,
- AL_FORMAT_STEREO_FLOAT32,
- mWindGen->windGenerate(mWindBuf,
- mWindBufSamples),
- mWindBufBytes,
- mWindBufFreq);
- error = alGetError();
- if(error != AL_NO_ERROR)
- {
- LL_WARNS() << "LLAudioEngine_OpenAL::updateWind() error swapping (bufferdata) buffers" << LL_ENDL;
- }
-
- alSourceQueueBuffers(mWindSource, 1, &buffer);
- error = alGetError();
- if(error != AL_NO_ERROR)
- {
- LL_WARNS() << "LLAudioEngine_OpenAL::updateWind() error swapping (queuing) buffers" << LL_ENDL;
- }
-
- --mNumEmptyWindALBuffers;
- }
-
- ALint playing;
- alGetSourcei(mWindSource, AL_SOURCE_STATE, &playing);
- if(playing != AL_PLAYING)
- {
- alSourcePlay(mWindSource);
-
- LL_DEBUGS() << "Wind had stopped - probably ran out of buffers - restarting: " << (unprocessed+mNumEmptyWindALBuffers) << " now queued." << LL_ENDL;
- }
-}
-
+/**
+ * @file audioengine_openal.cpp
+ * @brief implementation of audio engine using OpenAL
+ * support as a OpenAL 3D implementation
+ *
+ * $LicenseInfo:firstyear=2002&license=viewerlgpl$
+ * Second Life Viewer Source Code
+ * Copyright (C) 2010, Linden Research, Inc.
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation;
+ * version 2.1 of the License only.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with this library; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ *
+ * Linden Research, Inc., 945 Battery Street, San Francisco, CA 94111 USA
+ * $/LicenseInfo$
+ */
+
+#include "linden_common.h"
+#include "lldir.h"
+
+#include "llaudioengine_openal.h"
+#include "lllistener_openal.h"
+
+
+const float LLAudioEngine_OpenAL::WIND_BUFFER_SIZE_SEC = 0.05f;
+
+LLAudioEngine_OpenAL::LLAudioEngine_OpenAL()
+ :
+ mWindGen(NULL),
+ mWindBuf(NULL),
+ mWindBufFreq(0),
+ mWindBufSamples(0),
+ mWindBufBytes(0),
+ mWindSource(AL_NONE),
+ mNumEmptyWindALBuffers(MAX_NUM_WIND_BUFFERS)
+{
+}
+
+// virtual
+LLAudioEngine_OpenAL::~LLAudioEngine_OpenAL()
+{
+}
+
+// virtual
+bool LLAudioEngine_OpenAL::init(void* userdata, const std::string &app_title)
+{
+ mWindGen = NULL;
+ LLAudioEngine::init(userdata, app_title);
+
+ if(!alutInit(NULL, NULL))
+ {
+ LL_WARNS() << "LLAudioEngine_OpenAL::init() ALUT initialization failed: " << alutGetErrorString (alutGetError ()) << LL_ENDL;
+ return false;
+ }
+
+ LL_INFOS() << "LLAudioEngine_OpenAL::init() OpenAL successfully initialized" << LL_ENDL;
+
+ LL_INFOS() << "OpenAL version: "
+ << ll_safe_string(alGetString(AL_VERSION)) << LL_ENDL;
+ LL_INFOS() << "OpenAL vendor: "
+ << ll_safe_string(alGetString(AL_VENDOR)) << LL_ENDL;
+ LL_INFOS() << "OpenAL renderer: "
+ << ll_safe_string(alGetString(AL_RENDERER)) << LL_ENDL;
+
+ ALint major = alutGetMajorVersion ();
+ ALint minor = alutGetMinorVersion ();
+ LL_INFOS() << "ALUT version: " << major << "." << minor << LL_ENDL;
+
+ ALCdevice *device = alcGetContextsDevice(alcGetCurrentContext());
+
+ alcGetIntegerv(device, ALC_MAJOR_VERSION, 1, &major);
+ alcGetIntegerv(device, ALC_MAJOR_VERSION, 1, &minor);
+ LL_INFOS() << "ALC version: " << major << "." << minor << LL_ENDL;
+
+ LL_INFOS() << "ALC default device: "
+ << ll_safe_string(alcGetString(device,
+ ALC_DEFAULT_DEVICE_SPECIFIER))
+ << LL_ENDL;
+
+ return true;
+}
+
+// virtual
+std::string LLAudioEngine_OpenAL::getDriverName(bool verbose)
+{
+ ALCdevice *device = alcGetContextsDevice(alcGetCurrentContext());
+ std::ostringstream version;
+
+ version <<
+ "OpenAL";
+
+ if (verbose)
+ {
+ version <<
+ ", version " <<
+ ll_safe_string(alGetString(AL_VERSION)) <<
+ " / " <<
+ ll_safe_string(alGetString(AL_VENDOR)) <<
+ " / " <<
+ ll_safe_string(alGetString(AL_RENDERER));
+
+ if (device)
+ version <<
+ ": " <<
+ ll_safe_string(alcGetString(device,
+ ALC_DEFAULT_DEVICE_SPECIFIER));
+ }
+
+ return version.str();
+}
+
+// virtual
+void LLAudioEngine_OpenAL::allocateListener()
+{
+ mListenerp = (LLListener *) new LLListener_OpenAL();
+ if(!mListenerp)
+ {
+ LL_WARNS() << "LLAudioEngine_OpenAL::allocateListener() Listener creation failed" << LL_ENDL;
+ }
+}
+
+// virtual
+void LLAudioEngine_OpenAL::shutdown()
+{
+ LL_INFOS() << "About to LLAudioEngine::shutdown()" << LL_ENDL;
+ LLAudioEngine::shutdown();
+
+ // If a subsequent error occurs while there is still an error recorded
+ // internally, the second error will simply be ignored.
+ // Clear previous error to make sure we will captuare a valid failure reason
+ ALenum error = alutGetError();
+ if (error != ALUT_ERROR_NO_ERROR)
+ {
+ LL_WARNS() << "Uncleared error state prior to shutdown: "
+ << alutGetErrorString(error) << LL_ENDL;
+ }
+
+ LL_INFOS() << "About to alutExit()" << LL_ENDL;
+ if(!alutExit())
+ {
+ LL_WARNS() << "LLAudioEngine_OpenAL::shutdown() ALUT shutdown failed: " << alutGetErrorString (alutGetError ()) << LL_ENDL;
+ }
+
+ LL_INFOS() << "LLAudioEngine_OpenAL::shutdown() OpenAL successfully shut down" << LL_ENDL;
+
+ delete mListenerp;
+ mListenerp = NULL;
+}
+
+LLAudioBuffer *LLAudioEngine_OpenAL::createBuffer()
+{
+ return new LLAudioBufferOpenAL();
+}
+
+LLAudioChannel *LLAudioEngine_OpenAL::createChannel()
+{
+ return new LLAudioChannelOpenAL();
+}
+
+void LLAudioEngine_OpenAL::setInternalGain(F32 gain)
+{
+ //LL_INFOS() << "LLAudioEngine_OpenAL::setInternalGain() Gain: " << gain << LL_ENDL;
+ alListenerf(AL_GAIN, gain);
+}
+
+LLAudioChannelOpenAL::LLAudioChannelOpenAL()
+ :
+ mALSource(AL_NONE),
+ mLastSamplePos(0)
+{
+ alGenSources(1, &mALSource);
+}
+
+LLAudioChannelOpenAL::~LLAudioChannelOpenAL()
+{
+ cleanup();
+ alDeleteSources(1, &mALSource);
+}
+
+void LLAudioChannelOpenAL::cleanup()
+{
+ alSourceStop(mALSource);
+ mCurrentBufferp = NULL;
+}
+
+void LLAudioChannelOpenAL::play()
+{
+ if (mALSource == AL_NONE)
+ {
+ LL_WARNS() << "Playing without a mALSource, aborting" << LL_ENDL;
+ return;
+ }
+
+ if(!isPlaying())
+ {
+ alSourcePlay(mALSource);
+ getSource()->setPlayedOnce(true);
+ }
+}
+
+void LLAudioChannelOpenAL::playSynced(LLAudioChannel *channelp)
+{
+ if (channelp)
+ {
+ LLAudioChannelOpenAL *masterchannelp =
+ (LLAudioChannelOpenAL*)channelp;
+ if (mALSource != AL_NONE &&
+ masterchannelp->mALSource != AL_NONE)
+ {
+ // we have channels allocated to master and slave
+ ALfloat master_offset;
+ alGetSourcef(masterchannelp->mALSource, AL_SEC_OFFSET,
+ &master_offset);
+
+ LL_INFOS() << "Syncing with master at " << master_offset
+ << "sec" << LL_ENDL;
+ // *TODO: detect when this fails, maybe use AL_SAMPLE_
+ alSourcef(mALSource, AL_SEC_OFFSET, master_offset);
+ }
+ }
+ play();
+}
+
+bool LLAudioChannelOpenAL::isPlaying()
+{
+ if (mALSource != AL_NONE)
+ {
+ ALint state;
+ alGetSourcei(mALSource, AL_SOURCE_STATE, &state);
+ if(state == AL_PLAYING)
+ {
+ return true;
+ }
+ }
+
+ return false;
+}
+
+bool LLAudioChannelOpenAL::updateBuffer()
+{
+ if (!mCurrentSourcep)
+ {
+ // This channel isn't associated with any source, nothing
+ // to be updated
+ return false;
+ }
+
+ if (LLAudioChannel::updateBuffer())
+ {
+ // Base class update returned true, which means that we need to actually
+ // set up the source for a different buffer.
+ LLAudioBufferOpenAL *bufferp = (LLAudioBufferOpenAL *)mCurrentSourcep->getCurrentBuffer();
+ ALuint buffer = bufferp->getBuffer();
+ alSourcei(mALSource, AL_BUFFER, buffer);
+ mLastSamplePos = 0;
+ }
+
+ if (mCurrentSourcep)
+ {
+ alSourcef(mALSource, AL_GAIN,
+ mCurrentSourcep->getGain() * getSecondaryGain());
+ alSourcei(mALSource, AL_LOOPING,
+ mCurrentSourcep->isLoop() ? AL_TRUE : AL_FALSE);
+ alSourcef(mALSource, AL_ROLLOFF_FACTOR,
+ gAudiop->mListenerp->getRolloffFactor());
+ }
+
+ return true;
+}
+
+
+void LLAudioChannelOpenAL::updateLoop()
+{
+ if (mALSource == AL_NONE)
+ {
+ return;
+ }
+
+ // Hack: We keep track of whether we looped or not by seeing when the
+ // sample position looks like it's going backwards. Not reliable; may
+ // yield false negatives.
+ //
+ ALint cur_pos;
+ alGetSourcei(mALSource, AL_SAMPLE_OFFSET, &cur_pos);
+ if (cur_pos < mLastSamplePos)
+ {
+ mLoopedThisFrame = true;
+ }
+ mLastSamplePos = cur_pos;
+}
+
+
+void LLAudioChannelOpenAL::update3DPosition()
+{
+ if(!mCurrentSourcep)
+ {
+ return;
+ }
+ if (mCurrentSourcep->isForcedPriority())
+ {
+ alSource3f(mALSource, AL_POSITION, 0.0, 0.0, 0.0);
+ alSource3f(mALSource, AL_VELOCITY, 0.0, 0.0, 0.0);
+ alSourcei (mALSource, AL_SOURCE_RELATIVE, AL_TRUE);
+ } else {
+ LLVector3 float_pos;
+ float_pos.setVec(mCurrentSourcep->getPositionGlobal());
+ alSourcefv(mALSource, AL_POSITION, float_pos.mV);
+ alSourcefv(mALSource, AL_VELOCITY, mCurrentSourcep->getVelocity().mV);
+ alSourcei (mALSource, AL_SOURCE_RELATIVE, AL_FALSE);
+ }
+
+ alSourcef(mALSource, AL_GAIN, mCurrentSourcep->getGain() * getSecondaryGain());
+}
+
+LLAudioBufferOpenAL::LLAudioBufferOpenAL()
+{
+ mALBuffer = AL_NONE;
+}
+
+LLAudioBufferOpenAL::~LLAudioBufferOpenAL()
+{
+ cleanup();
+}
+
+void LLAudioBufferOpenAL::cleanup()
+{
+ if(mALBuffer != AL_NONE)
+ {
+ alDeleteBuffers(1, &mALBuffer);
+ mALBuffer = AL_NONE;
+ }
+}
+
+bool LLAudioBufferOpenAL::loadWAV(const std::string& filename)
+{
+ cleanup();
+ mALBuffer = alutCreateBufferFromFile(filename.c_str());
+ if(mALBuffer == AL_NONE)
+ {
+ ALenum error = alutGetError();
+ if (gDirUtilp->fileExists(filename))
+ {
+ LL_WARNS() <<
+ "LLAudioBufferOpenAL::loadWAV() Error loading "
+ << filename
+ << " " << alutGetErrorString(error) << LL_ENDL;
+ }
+ else
+ {
+ // It's common for the file to not actually exist.
+ LL_DEBUGS() <<
+ "LLAudioBufferOpenAL::loadWAV() Error loading "
+ << filename
+ << " " << alutGetErrorString(error) << LL_ENDL;
+ }
+ return false;
+ }
+
+ return true;
+}
+
+U32 LLAudioBufferOpenAL::getLength()
+{
+ if(mALBuffer == AL_NONE)
+ {
+ return 0;
+ }
+ ALint length;
+ alGetBufferi(mALBuffer, AL_SIZE, &length);
+ return length / 2; // convert size in bytes to size in (16-bit) samples
+}
+
+// ------------
+
+bool LLAudioEngine_OpenAL::initWind()
+{
+ ALenum error;
+ LL_INFOS() << "LLAudioEngine_OpenAL::initWind() start" << LL_ENDL;
+
+ mNumEmptyWindALBuffers = MAX_NUM_WIND_BUFFERS;
+
+ alGetError(); /* clear error */
+
+ alGenSources(1,&mWindSource);
+
+ if((error=alGetError()) != AL_NO_ERROR)
+ {
+ LL_WARNS() << "LLAudioEngine_OpenAL::initWind() Error creating wind sources: "<<error<<LL_ENDL;
+ }
+
+ mWindGen = new LLWindGen<WIND_SAMPLE_T>;
+
+ mWindBufFreq = mWindGen->getInputSamplingRate();
+ mWindBufSamples = llceil(mWindBufFreq * WIND_BUFFER_SIZE_SEC);
+ mWindBufBytes = mWindBufSamples * 2 /*stereo*/ * sizeof(WIND_SAMPLE_T);
+
+ mWindBuf = new WIND_SAMPLE_T [mWindBufSamples * 2 /*stereo*/];
+
+ if(mWindBuf==NULL)
+ {
+ LL_ERRS() << "LLAudioEngine_OpenAL::initWind() Error creating wind memory buffer" << LL_ENDL;
+ return false;
+ }
+
+ LL_INFOS() << "LLAudioEngine_OpenAL::initWind() done" << LL_ENDL;
+
+ return true;
+}
+
+void LLAudioEngine_OpenAL::cleanupWind()
+{
+ LL_INFOS() << "LLAudioEngine_OpenAL::cleanupWind()" << LL_ENDL;
+
+ if (mWindSource != AL_NONE)
+ {
+ // detach and delete all outstanding buffers on the wind source
+ alSourceStop(mWindSource);
+ ALint processed;
+ alGetSourcei(mWindSource, AL_BUFFERS_PROCESSED, &processed);
+ while (processed--)
+ {
+ ALuint buffer = AL_NONE;
+ alSourceUnqueueBuffers(mWindSource, 1, &buffer);
+ alDeleteBuffers(1, &buffer);
+ }
+
+ // delete the wind source itself
+ alDeleteSources(1, &mWindSource);
+
+ mWindSource = AL_NONE;
+ }
+
+ delete[] mWindBuf;
+ mWindBuf = NULL;
+
+ delete mWindGen;
+ mWindGen = NULL;
+}
+
+void LLAudioEngine_OpenAL::updateWind(LLVector3 wind_vec, F32 camera_altitude)
+{
+ LLVector3 wind_pos;
+ F64 pitch;
+ F64 center_freq;
+ ALenum error;
+
+ if (!mEnableWind)
+ return;
+
+ if(!mWindBuf)
+ return;
+
+ if (mWindUpdateTimer.checkExpirationAndReset(LL_WIND_UPDATE_INTERVAL))
+ {
+
+ // wind comes in as Linden coordinate (+X = forward, +Y = left, +Z = up)
+ // need to convert this to the conventional orientation DS3D and OpenAL use
+ // where +X = right, +Y = up, +Z = backwards
+
+ wind_vec.setVec(-wind_vec.mV[1], wind_vec.mV[2], -wind_vec.mV[0]);
+
+ pitch = 1.0 + mapWindVecToPitch(wind_vec);
+ center_freq = 80.0 * pow(pitch,2.5*(mapWindVecToGain(wind_vec)+1.0));
+
+ mWindGen->mTargetFreq = (F32)center_freq;
+ mWindGen->mTargetGain = (F32)mapWindVecToGain(wind_vec) * mMaxWindGain;
+ mWindGen->mTargetPanGainR = (F32)mapWindVecToPan(wind_vec);
+
+ alSourcei(mWindSource, AL_LOOPING, AL_FALSE);
+ alSource3f(mWindSource, AL_POSITION, 0.0, 0.0, 0.0);
+ alSource3f(mWindSource, AL_VELOCITY, 0.0, 0.0, 0.0);
+ alSourcef(mWindSource, AL_ROLLOFF_FACTOR, 0.0);
+ alSourcei(mWindSource, AL_SOURCE_RELATIVE, AL_TRUE);
+ }
+
+ // ok lets make a wind buffer now
+
+ ALint processed, queued, unprocessed;
+ alGetSourcei(mWindSource, AL_BUFFERS_PROCESSED, &processed);
+ alGetSourcei(mWindSource, AL_BUFFERS_QUEUED, &queued);
+ unprocessed = queued - processed;
+
+ // ensure that there are always at least 3x as many filled buffers
+ // queued as we managed to empty since last time.
+ mNumEmptyWindALBuffers = llmin(mNumEmptyWindALBuffers + processed * 3 - unprocessed, MAX_NUM_WIND_BUFFERS-unprocessed);
+ mNumEmptyWindALBuffers = llmax(mNumEmptyWindALBuffers, 0);
+
+ //LL_INFOS() << "mNumEmptyWindALBuffers: " << mNumEmptyWindALBuffers <<" (" << unprocessed << ":" << processed << ")" << LL_ENDL;
+
+ while(processed--) // unqueue old buffers
+ {
+ ALuint buffer;
+ ALenum error;
+ alGetError(); /* clear error */
+ alSourceUnqueueBuffers(mWindSource, 1, &buffer);
+ error = alGetError();
+ if(error != AL_NO_ERROR)
+ {
+ LL_WARNS() << "LLAudioEngine_OpenAL::updateWind() error swapping (unqueuing) buffers" << LL_ENDL;
+ }
+ else
+ {
+ alDeleteBuffers(1, &buffer);
+ }
+ }
+
+ unprocessed += mNumEmptyWindALBuffers;
+ while (mNumEmptyWindALBuffers > 0) // fill+queue new buffers
+ {
+ ALuint buffer;
+ alGetError(); /* clear error */
+ alGenBuffers(1,&buffer);
+ if((error=alGetError()) != AL_NO_ERROR)
+ {
+ LL_WARNS() << "LLAudioEngine_OpenAL::updateWind() Error creating wind buffer: " << error << LL_ENDL;
+ break;
+ }
+
+ alBufferData(buffer,
+ AL_FORMAT_STEREO_FLOAT32,
+ mWindGen->windGenerate(mWindBuf,
+ mWindBufSamples),
+ mWindBufBytes,
+ mWindBufFreq);
+ error = alGetError();
+ if(error != AL_NO_ERROR)
+ {
+ LL_WARNS() << "LLAudioEngine_OpenAL::updateWind() error swapping (bufferdata) buffers" << LL_ENDL;
+ }
+
+ alSourceQueueBuffers(mWindSource, 1, &buffer);
+ error = alGetError();
+ if(error != AL_NO_ERROR)
+ {
+ LL_WARNS() << "LLAudioEngine_OpenAL::updateWind() error swapping (queuing) buffers" << LL_ENDL;
+ }
+
+ --mNumEmptyWindALBuffers;
+ }
+
+ ALint playing;
+ alGetSourcei(mWindSource, AL_SOURCE_STATE, &playing);
+ if(playing != AL_PLAYING)
+ {
+ alSourcePlay(mWindSource);
+
+ LL_DEBUGS() << "Wind had stopped - probably ran out of buffers - restarting: " << (unprocessed+mNumEmptyWindALBuffers) << " now queued." << LL_ENDL;
+ }
+}
+
diff --git a/indra/llaudio/llaudioengine_openal.h b/indra/llaudio/llaudioengine_openal.h
index 6875bcc68b..574bec416d 100644
--- a/indra/llaudio/llaudioengine_openal.h
+++ b/indra/llaudio/llaudioengine_openal.h
@@ -1,108 +1,108 @@
-/**
- * @file audioengine_openal.cpp
- * @brief implementation of audio engine using OpenAL
- * support as a OpenAL 3D implementation
- *
- *
- * $LicenseInfo:firstyear=2002&license=viewerlgpl$
- * Second Life Viewer Source Code
- * Copyright (C) 2010, Linden Research, Inc.
- *
- * This library is free software; you can redistribute it and/or
- * modify it under the terms of the GNU Lesser General Public
- * License as published by the Free Software Foundation;
- * version 2.1 of the License only.
- *
- * This library is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
- * Lesser General Public License for more details.
- *
- * You should have received a copy of the GNU Lesser General Public
- * License along with this library; if not, write to the Free Software
- * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
- *
- * Linden Research, Inc., 945 Battery Street, San Francisco, CA 94111 USA
- * $/LicenseInfo$
- */
-
-
-#ifndef LL_AUDIOENGINE_OPENAL_H
-#define LL_AUDIOENGINE_OPENAL_H
-
-#include "llaudioengine.h"
-#include "lllistener_openal.h"
-#include "llwindgen.h"
-
-class LLAudioEngine_OpenAL : public LLAudioEngine
-{
- public:
- LLAudioEngine_OpenAL();
- virtual ~LLAudioEngine_OpenAL();
-
- virtual bool init(void *user_data, const std::string &app_title);
- virtual std::string getDriverName(bool verbose);
- virtual LLStreamingAudioInterface* createDefaultStreamingAudioImpl() const { return nullptr; }
- virtual void allocateListener();
-
- virtual void shutdown();
-
- void setInternalGain(F32 gain);
-
- LLAudioBuffer* createBuffer();
- LLAudioChannel* createChannel();
-
- /*virtual*/ bool initWind();
- /*virtual*/ void cleanupWind();
- /*virtual*/ void updateWind(LLVector3 direction, F32 camera_altitude);
-
- private:
- typedef F32 WIND_SAMPLE_T;
- LLWindGen<WIND_SAMPLE_T> *mWindGen;
- F32 *mWindBuf;
- U32 mWindBufFreq;
- U32 mWindBufSamples;
- U32 mWindBufBytes;
- ALuint mWindSource;
- int mNumEmptyWindALBuffers;
-
- static const int MAX_NUM_WIND_BUFFERS = 80;
- static const float WIND_BUFFER_SIZE_SEC; // 1/20th sec
-};
-
-class LLAudioChannelOpenAL : public LLAudioChannel
-{
- public:
- LLAudioChannelOpenAL();
- virtual ~LLAudioChannelOpenAL();
- protected:
- /*virtual*/ void play();
- /*virtual*/ void playSynced(LLAudioChannel *channelp);
- /*virtual*/ void cleanup();
- /*virtual*/ bool isPlaying();
-
- /*virtual*/ bool updateBuffer();
- /*virtual*/ void update3DPosition();
- /*virtual*/ void updateLoop();
-
- ALuint mALSource;
- ALint mLastSamplePos;
-};
-
-class LLAudioBufferOpenAL : public LLAudioBuffer{
- public:
- LLAudioBufferOpenAL();
- virtual ~LLAudioBufferOpenAL();
-
- bool loadWAV(const std::string& filename);
- U32 getLength();
-
- friend class LLAudioChannelOpenAL;
- protected:
- void cleanup();
- ALuint getBuffer() {return mALBuffer;}
-
- ALuint mALBuffer;
-};
-
-#endif
+/**
+ * @file audioengine_openal.cpp
+ * @brief implementation of audio engine using OpenAL
+ * support as a OpenAL 3D implementation
+ *
+ *
+ * $LicenseInfo:firstyear=2002&license=viewerlgpl$
+ * Second Life Viewer Source Code
+ * Copyright (C) 2010, Linden Research, Inc.
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation;
+ * version 2.1 of the License only.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with this library; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ *
+ * Linden Research, Inc., 945 Battery Street, San Francisco, CA 94111 USA
+ * $/LicenseInfo$
+ */
+
+
+#ifndef LL_AUDIOENGINE_OPENAL_H
+#define LL_AUDIOENGINE_OPENAL_H
+
+#include "llaudioengine.h"
+#include "lllistener_openal.h"
+#include "llwindgen.h"
+
+class LLAudioEngine_OpenAL : public LLAudioEngine
+{
+ public:
+ LLAudioEngine_OpenAL();
+ virtual ~LLAudioEngine_OpenAL();
+
+ virtual bool init(void *user_data, const std::string &app_title);
+ virtual std::string getDriverName(bool verbose);
+ virtual LLStreamingAudioInterface* createDefaultStreamingAudioImpl() const { return nullptr; }
+ virtual void allocateListener();
+
+ virtual void shutdown();
+
+ void setInternalGain(F32 gain);
+
+ LLAudioBuffer* createBuffer();
+ LLAudioChannel* createChannel();
+
+ /*virtual*/ bool initWind();
+ /*virtual*/ void cleanupWind();
+ /*virtual*/ void updateWind(LLVector3 direction, F32 camera_altitude);
+
+ private:
+ typedef F32 WIND_SAMPLE_T;
+ LLWindGen<WIND_SAMPLE_T> *mWindGen;
+ F32 *mWindBuf;
+ U32 mWindBufFreq;
+ U32 mWindBufSamples;
+ U32 mWindBufBytes;
+ ALuint mWindSource;
+ int mNumEmptyWindALBuffers;
+
+ static const int MAX_NUM_WIND_BUFFERS = 80;
+ static const float WIND_BUFFER_SIZE_SEC; // 1/20th sec
+};
+
+class LLAudioChannelOpenAL : public LLAudioChannel
+{
+ public:
+ LLAudioChannelOpenAL();
+ virtual ~LLAudioChannelOpenAL();
+ protected:
+ /*virtual*/ void play();
+ /*virtual*/ void playSynced(LLAudioChannel *channelp);
+ /*virtual*/ void cleanup();
+ /*virtual*/ bool isPlaying();
+
+ /*virtual*/ bool updateBuffer();
+ /*virtual*/ void update3DPosition();
+ /*virtual*/ void updateLoop();
+
+ ALuint mALSource;
+ ALint mLastSamplePos;
+};
+
+class LLAudioBufferOpenAL : public LLAudioBuffer{
+ public:
+ LLAudioBufferOpenAL();
+ virtual ~LLAudioBufferOpenAL();
+
+ bool loadWAV(const std::string& filename);
+ U32 getLength();
+
+ friend class LLAudioChannelOpenAL;
+ protected:
+ void cleanup();
+ ALuint getBuffer() {return mALBuffer;}
+
+ ALuint mALBuffer;
+};
+
+#endif
diff --git a/indra/llaudio/lllistener_openal.h b/indra/llaudio/lllistener_openal.h
index ca5eba8b4e..f1b69ddcef 100644
--- a/indra/llaudio/lllistener_openal.h
+++ b/indra/llaudio/lllistener_openal.h
@@ -1,59 +1,59 @@
-/**
- * @file listener_openal.h
- * @brief Description of LISTENER class abstracting the audio support
- * as an OpenAL implementation
- *
- * $LicenseInfo:firstyear=2000&license=viewerlgpl$
- * Second Life Viewer Source Code
- * Copyright (C) 2010, Linden Research, Inc.
- *
- * This library is free software; you can redistribute it and/or
- * modify it under the terms of the GNU Lesser General Public
- * License as published by the Free Software Foundation;
- * version 2.1 of the License only.
- *
- * This library is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
- * Lesser General Public License for more details.
- *
- * You should have received a copy of the GNU Lesser General Public
- * License along with this library; if not, write to the Free Software
- * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
- *
- * Linden Research, Inc., 945 Battery Street, San Francisco, CA 94111 USA
- * $/LicenseInfo$
- */
-
-#ifndef LL_LISTENER_OPENAL_H
-#define LL_LISTENER_OPENAL_H
-
-#include "lllistener.h"
-
-#include "AL/al.h"
-#include "AL/alut.h"
-#include "AL/alext.h"
-
-class LLListener_OpenAL : public LLListener
-{
- public:
- LLListener_OpenAL();
- virtual ~LLListener_OpenAL();
-
- virtual void translate(LLVector3 offset);
- virtual void setPosition(LLVector3 pos);
- virtual void setVelocity(LLVector3 vel);
- virtual void orient(LLVector3 up, LLVector3 at);
- virtual void commitDeferredChanges();
-
- virtual void setDopplerFactor(F32 factor);
- virtual F32 getDopplerFactor();
- virtual void setRolloffFactor(F32 factor);
- virtual F32 getRolloffFactor();
-
- protected:
- F32 mRolloffFactor;
-};
-
-#endif
-
+/**
+ * @file listener_openal.h
+ * @brief Description of LISTENER class abstracting the audio support
+ * as an OpenAL implementation
+ *
+ * $LicenseInfo:firstyear=2000&license=viewerlgpl$
+ * Second Life Viewer Source Code
+ * Copyright (C) 2010, Linden Research, Inc.
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation;
+ * version 2.1 of the License only.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with this library; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ *
+ * Linden Research, Inc., 945 Battery Street, San Francisco, CA 94111 USA
+ * $/LicenseInfo$
+ */
+
+#ifndef LL_LISTENER_OPENAL_H
+#define LL_LISTENER_OPENAL_H
+
+#include "lllistener.h"
+
+#include "AL/al.h"
+#include "AL/alut.h"
+#include "AL/alext.h"
+
+class LLListener_OpenAL : public LLListener
+{
+ public:
+ LLListener_OpenAL();
+ virtual ~LLListener_OpenAL();
+
+ virtual void translate(LLVector3 offset);
+ virtual void setPosition(LLVector3 pos);
+ virtual void setVelocity(LLVector3 vel);
+ virtual void orient(LLVector3 up, LLVector3 at);
+ virtual void commitDeferredChanges();
+
+ virtual void setDopplerFactor(F32 factor);
+ virtual F32 getDopplerFactor();
+ virtual void setRolloffFactor(F32 factor);
+ virtual F32 getRolloffFactor();
+
+ protected:
+ F32 mRolloffFactor;
+};
+
+#endif
+
diff --git a/indra/llaudio/llvorbisencode.cpp b/indra/llaudio/llvorbisencode.cpp
index 908175038e..83e7fad92f 100644
--- a/indra/llaudio/llvorbisencode.cpp
+++ b/indra/llaudio/llvorbisencode.cpp
@@ -1,506 +1,506 @@
-/**
- * @file vorbisencode.cpp
- * @brief Vorbis encoding routine routine for Indra.
- *
- * $LicenseInfo:firstyear=2000&license=viewerlgpl$
- * Second Life Viewer Source Code
- * Copyright (C) 2010, Linden Research, Inc.
- *
- * This library is free software; you can redistribute it and/or
- * modify it under the terms of the GNU Lesser General Public
- * License as published by the Free Software Foundation;
- * version 2.1 of the License only.
- *
- * This library is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
- * Lesser General Public License for more details.
- *
- * You should have received a copy of the GNU Lesser General Public
- * License along with this library; if not, write to the Free Software
- * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
- *
- * Linden Research, Inc., 945 Battery Street, San Francisco, CA 94111 USA
- * $/LicenseInfo$
- */
-
-#include "linden_common.h"
-
-#include "vorbis/vorbisenc.h"
-
-#include "llvorbisencode.h"
-#include "llerror.h"
-#include "llrand.h"
-#include "llmath.h"
-#include "llapr.h"
-
-//#if LL_DARWIN
-// MBW -- XXX -- Getting rid of SecondLifeVorbis for now
-#if 0
-#include "VorbisFramework.h"
-
-#define vorbis_analysis mac_vorbis_analysis
-#define vorbis_analysis_headerout mac_vorbis_analysis_headerout
-#define vorbis_analysis_init mac_vorbis_analysis_init
-#define vorbis_encode_ctl mac_vorbis_encode_ctl
-#define vorbis_encode_setup_init mac_vorbis_encode_setup_init
-#define vorbis_encode_setup_managed mac_vorbis_encode_setup_managed
-
-#define vorbis_info_init mac_vorbis_info_init
-#define vorbis_info_clear mac_vorbis_info_clear
-#define vorbis_comment_init mac_vorbis_comment_init
-#define vorbis_comment_clear mac_vorbis_comment_clear
-#define vorbis_block_init mac_vorbis_block_init
-#define vorbis_block_clear mac_vorbis_block_clear
-#define vorbis_dsp_clear mac_vorbis_dsp_clear
-#define vorbis_analysis_buffer mac_vorbis_analysis_buffer
-#define vorbis_analysis_wrote mac_vorbis_analysis_wrote
-#define vorbis_analysis_blockout mac_vorbis_analysis_blockout
-
-#define ogg_stream_packetin mac_ogg_stream_packetin
-#define ogg_stream_init mac_ogg_stream_init
-#define ogg_stream_flush mac_ogg_stream_flush
-#define ogg_stream_pageout mac_ogg_stream_pageout
-#define ogg_page_eos mac_ogg_page_eos
-#define ogg_stream_clear mac_ogg_stream_clear
-
-#endif
-
-S32 check_for_invalid_wav_formats(const std::string& in_fname, std::string& error_msg)
-{
- U16 num_channels = 0;
- U32 sample_rate = 0;
- U32 bits_per_sample = 0;
- U32 physical_file_size = 0;
- U32 chunk_length = 0;
- U32 raw_data_length = 0;
- U32 bytes_per_sec = 0;
- bool uncompressed_pcm = false;
-
- unsigned char wav_header[44]; /*Flawfinder: ignore*/
-
- error_msg.clear();
-
- //********************************
- LLAPRFile infile ;
- infile.open(in_fname,LL_APR_RB);
- //********************************
- if (!infile.getFileHandle())
- {
- error_msg = "CannotUploadSoundFile";
- return(LLVORBISENC_SOURCE_OPEN_ERR);
- }
-
- infile.read(wav_header, 44);
- physical_file_size = infile.seek(APR_END,0);
-
- if (strncmp((char *)&(wav_header[0]),"RIFF",4))
- {
- error_msg = "SoundFileNotRIFF";
- return(LLVORBISENC_WAV_FORMAT_ERR);
- }
-
- if (strncmp((char *)&(wav_header[8]),"WAVE",4))
- {
- error_msg = "SoundFileNotRIFF";
- return(LLVORBISENC_WAV_FORMAT_ERR);
- }
-
- // parse the chunks
-
- U32 file_pos = 12; // start at the first chunk (usually fmt but not always)
-
- while ((file_pos + 8)< physical_file_size)
- {
- infile.seek(APR_SET,file_pos);
- infile.read(wav_header, 44);
-
- chunk_length = ((U32) wav_header[7] << 24)
- + ((U32) wav_header[6] << 16)
- + ((U32) wav_header[5] << 8)
- + wav_header[4];
-
- if (chunk_length > physical_file_size - file_pos - 4)
- {
- infile.close();
- error_msg = "SoundFileInvalidChunkSize";
- return(LLVORBISENC_CHUNK_SIZE_ERR);
- }
-
-// LL_INFOS() << "chunk found: '" << wav_header[0] << wav_header[1] << wav_header[2] << wav_header[3] << "'" << LL_ENDL;
-
- if (!(strncmp((char *)&(wav_header[0]),"fmt ",4)))
- {
- if ((wav_header[8] == 0x01) && (wav_header[9] == 0x00))
- {
- uncompressed_pcm = true;
- }
- num_channels = ((U16) wav_header[11] << 8) + wav_header[10];
- sample_rate = ((U32) wav_header[15] << 24)
- + ((U32) wav_header[14] << 16)
- + ((U32) wav_header[13] << 8)
- + wav_header[12];
- bits_per_sample = ((U16) wav_header[23] << 8) + wav_header[22];
- bytes_per_sec = ((U32) wav_header[19] << 24)
- + ((U32) wav_header[18] << 16)
- + ((U32) wav_header[17] << 8)
- + wav_header[16];
- }
- else if (!(strncmp((char *)&(wav_header[0]),"data",4)))
- {
- raw_data_length = chunk_length;
- }
- file_pos += (chunk_length + 8);
- chunk_length = 0;
- }
- //****************
- infile.close();
- //****************
-
- if (!uncompressed_pcm)
- {
- error_msg = "SoundFileNotPCM";
- return(LLVORBISENC_PCM_FORMAT_ERR);
- }
-
- if ((num_channels < 1) || (num_channels > LLVORBIS_CLIP_MAX_CHANNELS))
- {
- error_msg = "SoundFileInvalidChannelCount";
- return(LLVORBISENC_MULTICHANNEL_ERR);
- }
-
- if (sample_rate != LLVORBIS_CLIP_SAMPLE_RATE)
- {
- error_msg = "SoundFileInvalidSampleRate";
- return(LLVORBISENC_UNSUPPORTED_SAMPLE_RATE);
- }
-
- if ((bits_per_sample != 16) && (bits_per_sample != 8))
- {
- error_msg = "SoundFileInvalidWordSize";
- return(LLVORBISENC_UNSUPPORTED_WORD_SIZE);
- }
-
- if (!raw_data_length)
- {
- error_msg = "SoundFileInvalidHeader";
- return(LLVORBISENC_CLIP_TOO_LONG);
- }
-
- F32 clip_length = (F32)raw_data_length/(F32)bytes_per_sec;
-
- if (clip_length > LLVORBIS_CLIP_MAX_TIME)
- {
- error_msg = "SoundFileInvalidTooLong";
- return(LLVORBISENC_CLIP_TOO_LONG);
- }
-
- return(LLVORBISENC_NOERR);
-}
-
-S32 encode_vorbis_file(const std::string& in_fname, const std::string& out_fname)
-{
-#define READ_BUFFER 1024
- unsigned char readbuffer[READ_BUFFER*4+44]; /* out of the data segment, not the stack */ /*Flawfinder: ignore*/
-
- ogg_stream_state os; /* take physical pages, weld into a logical stream of packets */
- ogg_page og; /* one Ogg bitstream page. Vorbis packets are inside */
- ogg_packet op; /* one raw packet of data for decode */
-
- vorbis_info vi; /* struct that stores all the static vorbis bitstream settings */
- vorbis_comment vc; /* struct that stores all the user comments */
-
- vorbis_dsp_state vd; /* central working state for the packet->PCM decoder */
- vorbis_block vb; /* local working space for packet->PCM decode */
-
- int eos=0;
- int result;
-
- U16 num_channels = 0;
- U32 sample_rate = 0;
- U32 bits_per_sample = 0;
-
- S32 format_error = 0;
- std::string error_msg;
- if ((format_error = check_for_invalid_wav_formats(in_fname, error_msg)))
- {
- LL_WARNS() << error_msg << ": " << in_fname << LL_ENDL;
- return(format_error);
- }
-
-#if 1
- unsigned char wav_header[44]; /*Flawfinder: ignore*/
-
- S32 data_left = 0;
-
- LLAPRFile infile ;
- infile.open(in_fname,LL_APR_RB);
- if (!infile.getFileHandle())
- {
- LL_WARNS() << "Couldn't open temporary ogg file for writing: " << in_fname
- << LL_ENDL;
- return(LLVORBISENC_SOURCE_OPEN_ERR);
- }
-
- LLAPRFile outfile ;
- outfile.open(out_fname,LL_APR_WPB);
- if (!outfile.getFileHandle())
- {
- LL_WARNS() << "Couldn't open upload sound file for reading: " << in_fname
- << LL_ENDL;
- return(LLVORBISENC_DEST_OPEN_ERR);
- }
-
- // parse the chunks
- U32 chunk_length = 0;
- U32 file_pos = 12; // start at the first chunk (usually fmt but not always)
-
- while (infile.eof() != APR_EOF)
- {
- infile.seek(APR_SET,file_pos);
- infile.read(wav_header, 44);
-
- chunk_length = ((U32) wav_header[7] << 24)
- + ((U32) wav_header[6] << 16)
- + ((U32) wav_header[5] << 8)
- + wav_header[4];
-
-// LL_INFOS() << "chunk found: '" << wav_header[0] << wav_header[1] << wav_header[2] << wav_header[3] << "'" << LL_ENDL;
-
- if (!(strncmp((char *)&(wav_header[0]),"fmt ",4)))
- {
- num_channels = ((U16) wav_header[11] << 8) + wav_header[10];
- sample_rate = ((U32) wav_header[15] << 24)
- + ((U32) wav_header[14] << 16)
- + ((U32) wav_header[13] << 8)
- + wav_header[12];
- bits_per_sample = ((U16) wav_header[23] << 8) + wav_header[22];
- }
- else if (!(strncmp((char *)&(wav_header[0]),"data",4)))
- {
- infile.seek(APR_SET,file_pos+8);
- // leave the file pointer at the beginning of the data chunk data
- data_left = chunk_length;
- break;
- }
- file_pos += (chunk_length + 8);
- chunk_length = 0;
- }
-
-
- /********** Encode setup ************/
-
- /* choose an encoding mode */
- /* (mode 0: 44kHz stereo uncoupled, roughly 128kbps VBR) */
- vorbis_info_init(&vi);
-
- // always encode to mono
-
- // SL-52913 & SL-53779 determined this quality level to be our 'good
- // enough' general-purpose quality level with a nice low bitrate.
- // Equivalent to oggenc -q0.5
- F32 quality = 0.05f;
-// quality = (bitrate==128000 ? 0.4f : 0.1);
-
-// if (vorbis_encode_init(&vi, /* num_channels */ 1 ,sample_rate, -1, bitrate, -1))
- if (vorbis_encode_init_vbr(&vi, /* num_channels */ 1 ,sample_rate, quality))
-// if (vorbis_encode_setup_managed(&vi,1,sample_rate,-1,bitrate,-1) ||
-// vorbis_encode_ctl(&vi,OV_ECTL_RATEMANAGE_AVG,NULL) ||
-// vorbis_encode_setup_init(&vi))
- {
- LL_WARNS() << "unable to initialize vorbis codec at quality " << quality << LL_ENDL;
- // LL_WARNS() << "unable to initialize vorbis codec at bitrate " << bitrate << LL_ENDL;
- return(LLVORBISENC_DEST_OPEN_ERR);
- }
-
- /* add a comment */
- vorbis_comment_init(&vc);
-// vorbis_comment_add(&vc,"Linden");
-
- /* set up the analysis state and auxiliary encoding storage */
- vorbis_analysis_init(&vd,&vi);
- vorbis_block_init(&vd,&vb);
-
- /* set up our packet->stream encoder */
- /* pick a random serial number; that way we can more likely build
- chained streams just by concatenation */
- ogg_stream_init(&os, ll_rand());
-
- /* Vorbis streams begin with three headers; the initial header (with
- most of the codec setup parameters) which is mandated by the Ogg
- bitstream spec. The second header holds any comment fields. The
- third header holds the bitstream codebook. We merely need to
- make the headers, then pass them to libvorbis one at a time;
- libvorbis handles the additional Ogg bitstream constraints */
-
- {
- ogg_packet header;
- ogg_packet header_comm;
- ogg_packet header_code;
-
- vorbis_analysis_headerout(&vd,&vc,&header,&header_comm,&header_code);
- ogg_stream_packetin(&os,&header); /* automatically placed in its own
- page */
- ogg_stream_packetin(&os,&header_comm);
- ogg_stream_packetin(&os,&header_code);
-
- /* We don't have to write out here, but doing so makes streaming
- * much easier, so we do, flushing ALL pages. This ensures the actual
- * audio data will start on a new page
- */
- while(!eos){
- int result=ogg_stream_flush(&os,&og);
- if(result==0)break;
- outfile.write(og.header, og.header_len);
- outfile.write(og.body, og.body_len);
- }
-
- }
-
-
- while(!eos)
- {
- long bytes_per_sample = bits_per_sample/8;
-
- long bytes=(long)infile.read(readbuffer,llclamp((S32)(READ_BUFFER*num_channels*bytes_per_sample),0,data_left)); /* stereo hardwired here */
-
- if (bytes==0)
- {
- /* end of file. this can be done implicitly in the mainline,
- but it's easier to see here in non-clever fashion.
- Tell the library we're at end of stream so that it can handle
- the last frame and mark end of stream in the output properly */
-
- vorbis_analysis_wrote(&vd,0);
-// eos = 1;
-
- }
- else
- {
- long i;
- long samples;
- int temp;
-
- data_left -= bytes;
- /* data to encode */
-
- /* expose the buffer to submit data */
- float **buffer=vorbis_analysis_buffer(&vd,READ_BUFFER);
-
- i = 0;
- samples = bytes / (num_channels * bytes_per_sample);
-
- if (num_channels == 2)
- {
- if (bytes_per_sample == 2)
- {
- /* uninterleave samples */
- for(i=0; i<samples ;i++)
- {
- temp = ((signed char *)readbuffer)[i*4+1]; /*Flawfinder: ignore*/
- temp += ((signed char *)readbuffer)[i*4+3]; /*Flawfinder: ignore*/
- temp <<= 8;
- temp += readbuffer[i*4];
- temp += readbuffer[i*4+2];
-
- buffer[0][i] = ((float)temp) / 65536.f;
- }
- }
- else // presume it's 1 byte per which is unsigned (F#@%ing wav "standard")
- {
- /* uninterleave samples */
- for(i=0; i<samples ;i++)
- {
- temp = readbuffer[i*2+0];
- temp += readbuffer[i*2+1];
- temp -= 256;
- buffer[0][i] = ((float)temp) / 256.f;
- }
- }
- }
- else if (num_channels == 1)
- {
- if (bytes_per_sample == 2)
- {
- for(i=0; i < samples ;i++)
- {
- temp = ((signed char*)readbuffer)[i*2+1];
- temp <<= 8;
- temp += readbuffer[i*2];
- buffer[0][i] = ((float)temp) / 32768.f;
- }
- }
- else // presume it's 1 byte per which is unsigned (F#@%ing wav "standard")
- {
- for(i=0; i < samples ;i++)
- {
- temp = readbuffer[i];
- temp -= 128;
- buffer[0][i] = ((float)temp) / 128.f;
- }
- }
- }
-
- /* tell the library how much we actually submitted */
- vorbis_analysis_wrote(&vd,i);
- }
-
- /* vorbis does some data preanalysis, then divvies up blocks for
- more involved (potentially parallel) processing. Get a single
- block for encoding now */
- while(vorbis_analysis_blockout(&vd,&vb)==1)
- {
-
- /* analysis */
- /* Do the main analysis, creating a packet */
- vorbis_analysis(&vb, NULL);
- vorbis_bitrate_addblock(&vb);
-
- while(vorbis_bitrate_flushpacket(&vd, &op))
- {
-
- /* weld the packet into the bitstream */
- ogg_stream_packetin(&os,&op);
-
- /* write out pages (if any) */
- while(!eos)
- {
- result = ogg_stream_pageout(&os,&og);
-
- if(result==0)
- break;
-
- outfile.write(og.header, og.header_len);
- outfile.write(og.body, og.body_len);
-
- /* this could be set above, but for illustrative purposes, I do
- it here (to show that vorbis does know where the stream ends) */
-
- if(ogg_page_eos(&og))
- eos=1;
-
- }
- }
- }
- }
-
-
-
- /* clean up and exit. vorbis_info_clear() must be called last */
-
- ogg_stream_clear(&os);
- vorbis_block_clear(&vb);
- vorbis_dsp_clear(&vd);
- vorbis_comment_clear(&vc);
- vorbis_info_clear(&vi);
-
- /* ogg_page and ogg_packet structs always point to storage in
- libvorbis. They're never freed or manipulated directly */
-
-// fprintf(stderr,"Vorbis encoding: Done.\n");
- LL_INFOS() << "Vorbis encoding: Done." << LL_ENDL;
-
-#endif
- return(LLVORBISENC_NOERR);
-
-}
+/**
+ * @file vorbisencode.cpp
+ * @brief Vorbis encoding routine routine for Indra.
+ *
+ * $LicenseInfo:firstyear=2000&license=viewerlgpl$
+ * Second Life Viewer Source Code
+ * Copyright (C) 2010, Linden Research, Inc.
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation;
+ * version 2.1 of the License only.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with this library; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ *
+ * Linden Research, Inc., 945 Battery Street, San Francisco, CA 94111 USA
+ * $/LicenseInfo$
+ */
+
+#include "linden_common.h"
+
+#include "vorbis/vorbisenc.h"
+
+#include "llvorbisencode.h"
+#include "llerror.h"
+#include "llrand.h"
+#include "llmath.h"
+#include "llapr.h"
+
+//#if LL_DARWIN
+// MBW -- XXX -- Getting rid of SecondLifeVorbis for now
+#if 0
+#include "VorbisFramework.h"
+
+#define vorbis_analysis mac_vorbis_analysis
+#define vorbis_analysis_headerout mac_vorbis_analysis_headerout
+#define vorbis_analysis_init mac_vorbis_analysis_init
+#define vorbis_encode_ctl mac_vorbis_encode_ctl
+#define vorbis_encode_setup_init mac_vorbis_encode_setup_init
+#define vorbis_encode_setup_managed mac_vorbis_encode_setup_managed
+
+#define vorbis_info_init mac_vorbis_info_init
+#define vorbis_info_clear mac_vorbis_info_clear
+#define vorbis_comment_init mac_vorbis_comment_init
+#define vorbis_comment_clear mac_vorbis_comment_clear
+#define vorbis_block_init mac_vorbis_block_init
+#define vorbis_block_clear mac_vorbis_block_clear
+#define vorbis_dsp_clear mac_vorbis_dsp_clear
+#define vorbis_analysis_buffer mac_vorbis_analysis_buffer
+#define vorbis_analysis_wrote mac_vorbis_analysis_wrote
+#define vorbis_analysis_blockout mac_vorbis_analysis_blockout
+
+#define ogg_stream_packetin mac_ogg_stream_packetin
+#define ogg_stream_init mac_ogg_stream_init
+#define ogg_stream_flush mac_ogg_stream_flush
+#define ogg_stream_pageout mac_ogg_stream_pageout
+#define ogg_page_eos mac_ogg_page_eos
+#define ogg_stream_clear mac_ogg_stream_clear
+
+#endif
+
+S32 check_for_invalid_wav_formats(const std::string& in_fname, std::string& error_msg)
+{
+ U16 num_channels = 0;
+ U32 sample_rate = 0;
+ U32 bits_per_sample = 0;
+ U32 physical_file_size = 0;
+ U32 chunk_length = 0;
+ U32 raw_data_length = 0;
+ U32 bytes_per_sec = 0;
+ bool uncompressed_pcm = false;
+
+ unsigned char wav_header[44]; /*Flawfinder: ignore*/
+
+ error_msg.clear();
+
+ //********************************
+ LLAPRFile infile ;
+ infile.open(in_fname,LL_APR_RB);
+ //********************************
+ if (!infile.getFileHandle())
+ {
+ error_msg = "CannotUploadSoundFile";
+ return(LLVORBISENC_SOURCE_OPEN_ERR);
+ }
+
+ infile.read(wav_header, 44);
+ physical_file_size = infile.seek(APR_END,0);
+
+ if (strncmp((char *)&(wav_header[0]),"RIFF",4))
+ {
+ error_msg = "SoundFileNotRIFF";
+ return(LLVORBISENC_WAV_FORMAT_ERR);
+ }
+
+ if (strncmp((char *)&(wav_header[8]),"WAVE",4))
+ {
+ error_msg = "SoundFileNotRIFF";
+ return(LLVORBISENC_WAV_FORMAT_ERR);
+ }
+
+ // parse the chunks
+
+ U32 file_pos = 12; // start at the first chunk (usually fmt but not always)
+
+ while ((file_pos + 8)< physical_file_size)
+ {
+ infile.seek(APR_SET,file_pos);
+ infile.read(wav_header, 44);
+
+ chunk_length = ((U32) wav_header[7] << 24)
+ + ((U32) wav_header[6] << 16)
+ + ((U32) wav_header[5] << 8)
+ + wav_header[4];
+
+ if (chunk_length > physical_file_size - file_pos - 4)
+ {
+ infile.close();
+ error_msg = "SoundFileInvalidChunkSize";
+ return(LLVORBISENC_CHUNK_SIZE_ERR);
+ }
+
+// LL_INFOS() << "chunk found: '" << wav_header[0] << wav_header[1] << wav_header[2] << wav_header[3] << "'" << LL_ENDL;
+
+ if (!(strncmp((char *)&(wav_header[0]),"fmt ",4)))
+ {
+ if ((wav_header[8] == 0x01) && (wav_header[9] == 0x00))
+ {
+ uncompressed_pcm = true;
+ }
+ num_channels = ((U16) wav_header[11] << 8) + wav_header[10];
+ sample_rate = ((U32) wav_header[15] << 24)
+ + ((U32) wav_header[14] << 16)
+ + ((U32) wav_header[13] << 8)
+ + wav_header[12];
+ bits_per_sample = ((U16) wav_header[23] << 8) + wav_header[22];
+ bytes_per_sec = ((U32) wav_header[19] << 24)
+ + ((U32) wav_header[18] << 16)
+ + ((U32) wav_header[17] << 8)
+ + wav_header[16];
+ }
+ else if (!(strncmp((char *)&(wav_header[0]),"data",4)))
+ {
+ raw_data_length = chunk_length;
+ }
+ file_pos += (chunk_length + 8);
+ chunk_length = 0;
+ }
+ //****************
+ infile.close();
+ //****************
+
+ if (!uncompressed_pcm)
+ {
+ error_msg = "SoundFileNotPCM";
+ return(LLVORBISENC_PCM_FORMAT_ERR);
+ }
+
+ if ((num_channels < 1) || (num_channels > LLVORBIS_CLIP_MAX_CHANNELS))
+ {
+ error_msg = "SoundFileInvalidChannelCount";
+ return(LLVORBISENC_MULTICHANNEL_ERR);
+ }
+
+ if (sample_rate != LLVORBIS_CLIP_SAMPLE_RATE)
+ {
+ error_msg = "SoundFileInvalidSampleRate";
+ return(LLVORBISENC_UNSUPPORTED_SAMPLE_RATE);
+ }
+
+ if ((bits_per_sample != 16) && (bits_per_sample != 8))
+ {
+ error_msg = "SoundFileInvalidWordSize";
+ return(LLVORBISENC_UNSUPPORTED_WORD_SIZE);
+ }
+
+ if (!raw_data_length)
+ {
+ error_msg = "SoundFileInvalidHeader";
+ return(LLVORBISENC_CLIP_TOO_LONG);
+ }
+
+ F32 clip_length = (F32)raw_data_length/(F32)bytes_per_sec;
+
+ if (clip_length > LLVORBIS_CLIP_MAX_TIME)
+ {
+ error_msg = "SoundFileInvalidTooLong";
+ return(LLVORBISENC_CLIP_TOO_LONG);
+ }
+
+ return(LLVORBISENC_NOERR);
+}
+
+S32 encode_vorbis_file(const std::string& in_fname, const std::string& out_fname)
+{
+#define READ_BUFFER 1024
+ unsigned char readbuffer[READ_BUFFER*4+44]; /* out of the data segment, not the stack */ /*Flawfinder: ignore*/
+
+ ogg_stream_state os; /* take physical pages, weld into a logical stream of packets */
+ ogg_page og; /* one Ogg bitstream page. Vorbis packets are inside */
+ ogg_packet op; /* one raw packet of data for decode */
+
+ vorbis_info vi; /* struct that stores all the static vorbis bitstream settings */
+ vorbis_comment vc; /* struct that stores all the user comments */
+
+ vorbis_dsp_state vd; /* central working state for the packet->PCM decoder */
+ vorbis_block vb; /* local working space for packet->PCM decode */
+
+ int eos=0;
+ int result;
+
+ U16 num_channels = 0;
+ U32 sample_rate = 0;
+ U32 bits_per_sample = 0;
+
+ S32 format_error = 0;
+ std::string error_msg;
+ if ((format_error = check_for_invalid_wav_formats(in_fname, error_msg)))
+ {
+ LL_WARNS() << error_msg << ": " << in_fname << LL_ENDL;
+ return(format_error);
+ }
+
+#if 1
+ unsigned char wav_header[44]; /*Flawfinder: ignore*/
+
+ S32 data_left = 0;
+
+ LLAPRFile infile ;
+ infile.open(in_fname,LL_APR_RB);
+ if (!infile.getFileHandle())
+ {
+ LL_WARNS() << "Couldn't open temporary ogg file for writing: " << in_fname
+ << LL_ENDL;
+ return(LLVORBISENC_SOURCE_OPEN_ERR);
+ }
+
+ LLAPRFile outfile ;
+ outfile.open(out_fname,LL_APR_WPB);
+ if (!outfile.getFileHandle())
+ {
+ LL_WARNS() << "Couldn't open upload sound file for reading: " << in_fname
+ << LL_ENDL;
+ return(LLVORBISENC_DEST_OPEN_ERR);
+ }
+
+ // parse the chunks
+ U32 chunk_length = 0;
+ U32 file_pos = 12; // start at the first chunk (usually fmt but not always)
+
+ while (infile.eof() != APR_EOF)
+ {
+ infile.seek(APR_SET,file_pos);
+ infile.read(wav_header, 44);
+
+ chunk_length = ((U32) wav_header[7] << 24)
+ + ((U32) wav_header[6] << 16)
+ + ((U32) wav_header[5] << 8)
+ + wav_header[4];
+
+// LL_INFOS() << "chunk found: '" << wav_header[0] << wav_header[1] << wav_header[2] << wav_header[3] << "'" << LL_ENDL;
+
+ if (!(strncmp((char *)&(wav_header[0]),"fmt ",4)))
+ {
+ num_channels = ((U16) wav_header[11] << 8) + wav_header[10];
+ sample_rate = ((U32) wav_header[15] << 24)
+ + ((U32) wav_header[14] << 16)
+ + ((U32) wav_header[13] << 8)
+ + wav_header[12];
+ bits_per_sample = ((U16) wav_header[23] << 8) + wav_header[22];
+ }
+ else if (!(strncmp((char *)&(wav_header[0]),"data",4)))
+ {
+ infile.seek(APR_SET,file_pos+8);
+ // leave the file pointer at the beginning of the data chunk data
+ data_left = chunk_length;
+ break;
+ }
+ file_pos += (chunk_length + 8);
+ chunk_length = 0;
+ }
+
+
+ /********** Encode setup ************/
+
+ /* choose an encoding mode */
+ /* (mode 0: 44kHz stereo uncoupled, roughly 128kbps VBR) */
+ vorbis_info_init(&vi);
+
+ // always encode to mono
+
+ // SL-52913 & SL-53779 determined this quality level to be our 'good
+ // enough' general-purpose quality level with a nice low bitrate.
+ // Equivalent to oggenc -q0.5
+ F32 quality = 0.05f;
+// quality = (bitrate==128000 ? 0.4f : 0.1);
+
+// if (vorbis_encode_init(&vi, /* num_channels */ 1 ,sample_rate, -1, bitrate, -1))
+ if (vorbis_encode_init_vbr(&vi, /* num_channels */ 1 ,sample_rate, quality))
+// if (vorbis_encode_setup_managed(&vi,1,sample_rate,-1,bitrate,-1) ||
+// vorbis_encode_ctl(&vi,OV_ECTL_RATEMANAGE_AVG,NULL) ||
+// vorbis_encode_setup_init(&vi))
+ {
+ LL_WARNS() << "unable to initialize vorbis codec at quality " << quality << LL_ENDL;
+ // LL_WARNS() << "unable to initialize vorbis codec at bitrate " << bitrate << LL_ENDL;
+ return(LLVORBISENC_DEST_OPEN_ERR);
+ }
+
+ /* add a comment */
+ vorbis_comment_init(&vc);
+// vorbis_comment_add(&vc,"Linden");
+
+ /* set up the analysis state and auxiliary encoding storage */
+ vorbis_analysis_init(&vd,&vi);
+ vorbis_block_init(&vd,&vb);
+
+ /* set up our packet->stream encoder */
+ /* pick a random serial number; that way we can more likely build
+ chained streams just by concatenation */
+ ogg_stream_init(&os, ll_rand());
+
+ /* Vorbis streams begin with three headers; the initial header (with
+ most of the codec setup parameters) which is mandated by the Ogg
+ bitstream spec. The second header holds any comment fields. The
+ third header holds the bitstream codebook. We merely need to
+ make the headers, then pass them to libvorbis one at a time;
+ libvorbis handles the additional Ogg bitstream constraints */
+
+ {
+ ogg_packet header;
+ ogg_packet header_comm;
+ ogg_packet header_code;
+
+ vorbis_analysis_headerout(&vd,&vc,&header,&header_comm,&header_code);
+ ogg_stream_packetin(&os,&header); /* automatically placed in its own
+ page */
+ ogg_stream_packetin(&os,&header_comm);
+ ogg_stream_packetin(&os,&header_code);
+
+ /* We don't have to write out here, but doing so makes streaming
+ * much easier, so we do, flushing ALL pages. This ensures the actual
+ * audio data will start on a new page
+ */
+ while(!eos){
+ int result=ogg_stream_flush(&os,&og);
+ if(result==0)break;
+ outfile.write(og.header, og.header_len);
+ outfile.write(og.body, og.body_len);
+ }
+
+ }
+
+
+ while(!eos)
+ {
+ long bytes_per_sample = bits_per_sample/8;
+
+ long bytes=(long)infile.read(readbuffer,llclamp((S32)(READ_BUFFER*num_channels*bytes_per_sample),0,data_left)); /* stereo hardwired here */
+
+ if (bytes==0)
+ {
+ /* end of file. this can be done implicitly in the mainline,
+ but it's easier to see here in non-clever fashion.
+ Tell the library we're at end of stream so that it can handle
+ the last frame and mark end of stream in the output properly */
+
+ vorbis_analysis_wrote(&vd,0);
+// eos = 1;
+
+ }
+ else
+ {
+ long i;
+ long samples;
+ int temp;
+
+ data_left -= bytes;
+ /* data to encode */
+
+ /* expose the buffer to submit data */
+ float **buffer=vorbis_analysis_buffer(&vd,READ_BUFFER);
+
+ i = 0;
+ samples = bytes / (num_channels * bytes_per_sample);
+
+ if (num_channels == 2)
+ {
+ if (bytes_per_sample == 2)
+ {
+ /* uninterleave samples */
+ for(i=0; i<samples ;i++)
+ {
+ temp = ((signed char *)readbuffer)[i*4+1]; /*Flawfinder: ignore*/
+ temp += ((signed char *)readbuffer)[i*4+3]; /*Flawfinder: ignore*/
+ temp <<= 8;
+ temp += readbuffer[i*4];
+ temp += readbuffer[i*4+2];
+
+ buffer[0][i] = ((float)temp) / 65536.f;
+ }
+ }
+ else // presume it's 1 byte per which is unsigned (F#@%ing wav "standard")
+ {
+ /* uninterleave samples */
+ for(i=0; i<samples ;i++)
+ {
+ temp = readbuffer[i*2+0];
+ temp += readbuffer[i*2+1];
+ temp -= 256;
+ buffer[0][i] = ((float)temp) / 256.f;
+ }
+ }
+ }
+ else if (num_channels == 1)
+ {
+ if (bytes_per_sample == 2)
+ {
+ for(i=0; i < samples ;i++)
+ {
+ temp = ((signed char*)readbuffer)[i*2+1];
+ temp <<= 8;
+ temp += readbuffer[i*2];
+ buffer[0][i] = ((float)temp) / 32768.f;
+ }
+ }
+ else // presume it's 1 byte per which is unsigned (F#@%ing wav "standard")
+ {
+ for(i=0; i < samples ;i++)
+ {
+ temp = readbuffer[i];
+ temp -= 128;
+ buffer[0][i] = ((float)temp) / 128.f;
+ }
+ }
+ }
+
+ /* tell the library how much we actually submitted */
+ vorbis_analysis_wrote(&vd,i);
+ }
+
+ /* vorbis does some data preanalysis, then divvies up blocks for
+ more involved (potentially parallel) processing. Get a single
+ block for encoding now */
+ while(vorbis_analysis_blockout(&vd,&vb)==1)
+ {
+
+ /* analysis */
+ /* Do the main analysis, creating a packet */
+ vorbis_analysis(&vb, NULL);
+ vorbis_bitrate_addblock(&vb);
+
+ while(vorbis_bitrate_flushpacket(&vd, &op))
+ {
+
+ /* weld the packet into the bitstream */
+ ogg_stream_packetin(&os,&op);
+
+ /* write out pages (if any) */
+ while(!eos)
+ {
+ result = ogg_stream_pageout(&os,&og);
+
+ if(result==0)
+ break;
+
+ outfile.write(og.header, og.header_len);
+ outfile.write(og.body, og.body_len);
+
+ /* this could be set above, but for illustrative purposes, I do
+ it here (to show that vorbis does know where the stream ends) */
+
+ if(ogg_page_eos(&og))
+ eos=1;
+
+ }
+ }
+ }
+ }
+
+
+
+ /* clean up and exit. vorbis_info_clear() must be called last */
+
+ ogg_stream_clear(&os);
+ vorbis_block_clear(&vb);
+ vorbis_dsp_clear(&vd);
+ vorbis_comment_clear(&vc);
+ vorbis_info_clear(&vi);
+
+ /* ogg_page and ogg_packet structs always point to storage in
+ libvorbis. They're never freed or manipulated directly */
+
+// fprintf(stderr,"Vorbis encoding: Done.\n");
+ LL_INFOS() << "Vorbis encoding: Done." << LL_ENDL;
+
+#endif
+ return(LLVORBISENC_NOERR);
+
+}
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