svn merge -r 129841:129910 svn+ssh://svn.lindenlab.com/svn/linden/branches/moss/pluginapi_05-merge@129910

svn merge -r 129913:131718 svn+ssh://svn.lindenlab.com/svn/linden/branches/pluginapi/pluginapi_05

Some branch shenannigans in the pluginapi_05 branch caused this to become a two-part merge.

1 files changed, 505 insertions, 0 deletions

diff --git a/indra/llaudio/llvorbisencode.cpp b/indra/llaudio/llvorbisencode.cpp
new file mode 100644
index 0000000000..8ee082a245
--- /dev/null
+++ b/indra/llaudio/llvorbisencode.cpp
@@ -0,0 +1,505 @@
+/** 
+ * @file vorbisencode.cpp
+ * @brief Vorbis encoding routine routine for Indra.
+ *
+ * $LicenseInfo:firstyear=2000&license=viewergpl$
+ * 
+ * Copyright (c) 2000-2009, Linden Research, Inc.
+ * 
+ * Second Life Viewer Source Code
+ * The source code in this file ("Source Code") is provided by Linden Lab
+ * to you under the terms of the GNU General Public License, version 2.0
+ * ("GPL"), unless you have obtained a separate licensing agreement
+ * ("Other License"), formally executed by you and Linden Lab.  Terms of
+ * the GPL can be found in doc/GPL-license.txt in this distribution, or
+ * online at http://secondlifegrid.net/programs/open_source/licensing/gplv2
+ * 
+ * There are special exceptions to the terms and conditions of the GPL as
+ * it is applied to this Source Code. View the full text of the exception
+ * in the file doc/FLOSS-exception.txt in this software distribution, or
+ * online at
+ * http://secondlifegrid.net/programs/open_source/licensing/flossexception
+ * 
+ * By copying, modifying or distributing this software, you acknowledge
+ * that you have read and understood your obligations described above,
+ * and agree to abide by those obligations.
+ * 
+ * ALL LINDEN LAB SOURCE CODE IS PROVIDED "AS IS." LINDEN LAB MAKES NO
+ * WARRANTIES, EXPRESS, IMPLIED OR OTHERWISE, REGARDING ITS ACCURACY,
+ * COMPLETENESS OR PERFORMANCE.
+ * $/LicenseInfo$
+ */
+
+#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 -- no fmod means no name collisions.
+#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];
+
+//		llinfos << "chunk found: '" << wav_header[0] << wav_header[1] << wav_header[2] << wav_header[3] << "'" << llendl;
+
+		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 > 2))
+	{	
+		error_msg = "SoundFileInvalidChannelCount";
+		return(LLVORBISENC_MULTICHANNEL_ERR);
+	}
+
+	if (sample_rate != 44100)
+	{	
+		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 > 10.0f)
+	{
+		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)))
+	{
+		llwarns << error_msg << ": " << in_fname << llendl;
+		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())
+	{
+		llwarns << "Couldn't open temporary ogg file for writing: " << in_fname
+			<< llendl;
+		return(LLVORBISENC_SOURCE_OPEN_ERR);
+	}
+
+	LLAPRFile outfile ;
+	outfile.open(out_fname,LL_APR_WPB);
+	if (!outfile.getFileHandle())
+	{
+		llwarns << "Couldn't open upload sound file for reading: " << in_fname
+			<< llendl;
+		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];
+		 
+//		 llinfos << "chunk found: '" << wav_header[0] << wav_header[1] << wav_header[2] << wav_header[3] << "'" << llendl;
+		 
+		 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))
+	{
+		llwarns << "unable to initialize vorbis codec at quality " << quality << llendl;
+		//		llwarns << "unable to initialize vorbis codec at bitrate " << bitrate << llendl;
+		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");
+	 llinfos << "Vorbis encoding: Done." << llendl;
+	 
+#endif
+	 return(LLVORBISENC_NOERR);
+	 
+}