1 /* 2 * CDDL HEADER START 3 * 4 * The contents of this file are subject to the terms of the 5 * Common Development and Distribution License, Version 1.0 only 6 * (the "License"). You may not use this file except in compliance 7 * with the License. 8 * 9 * You can obtain a copy of the license at usr/src/OPENSOLARIS.LICENSE 10 * or http://www.opensolaris.org/os/licensing. 11 * See the License for the specific language governing permissions 12 * and limitations under the License. 13 * 14 * When distributing Covered Code, include this CDDL HEADER in each 15 * file and include the License file at usr/src/OPENSOLARIS.LICENSE. 16 * If applicable, add the following below this CDDL HEADER, with the 17 * fields enclosed by brackets "[]" replaced with your own identifying 18 * information: Portions Copyright [yyyy] [name of copyright owner] 19 * 20 * CDDL HEADER END 21 */ 22 /* 23 * Copyright (c) 1992-2001 by Sun Microsystems, Inc. 24 * All rights reserved. 25 */ 26 27 #pragma ident "%Z%%M% %I% %E% SMI" 28 29 #include <stdlib.h> 30 #include <memory.h> 31 #include <math.h> 32 33 #include <AudioDebug.h> 34 #include <AudioTypeSampleRate.h> 35 36 // This is the first stab at a conversion class for Sample Rate conversions 37 38 // class AudioTypeSampleRate methods 39 40 // Constructor 41 AudioTypeSampleRate:: 42 AudioTypeSampleRate(int inrate, int outrate) : 43 resampler(inrate, outrate), input_rate(inrate), output_rate(outrate) 44 { 45 } 46 47 // Destructor 48 AudioTypeSampleRate:: 49 ~AudioTypeSampleRate() 50 { 51 } 52 53 // Test conversion possibilities. 54 // Return TRUE if conversion to/from the specified type is possible. 55 Boolean AudioTypeSampleRate:: 56 CanConvert( 57 AudioHdr h) const // target header 58 { 59 if ((input_rate <= 0) || (output_rate <= 0)) 60 return (FALSE); 61 if ((h.encoding != LINEAR) || 62 ((h.sample_rate != output_rate) && (h.sample_rate != input_rate)) || 63 (h.bytes_per_unit != 2) || 64 (h.channels != 1)) { 65 return (FALSE); 66 } 67 return (TRUE); 68 } 69 70 71 // Convert buffer to the specified type 72 // May replace the buffer with a new one, if necessary 73 AudioError AudioTypeSampleRate:: 74 Convert( 75 AudioBuffer*& inbuf, // data buffer to process 76 AudioHdr outhdr) // target header 77 { 78 AudioBuffer* outbuf; 79 AudioHdr inhdr; 80 Double length; 81 int i; 82 size_t nsamps; 83 size_t insamps; 84 AudioError err; 85 86 inhdr = inbuf->GetHeader(); 87 length = inbuf->GetLength(); 88 89 if (Undefined(length)) { 90 return (AUDIO_ERR_BADARG); 91 } 92 93 // Make sure we're not being asked to do the impossible 94 // XXX - need a better error code 95 if ((err = inhdr.Validate()) || (err = outhdr.Validate())) { 96 return (err); 97 } 98 99 // If the requested conversion is different than what was initially 100 // established, then return an error. 101 // XXX - Maybe one day flush and re-init the filter 102 if ((inhdr.sample_rate != input_rate) || 103 (outhdr.sample_rate != output_rate)) { 104 return (AUDIO_ERR_BADARG); 105 } 106 107 // If conversion is a no-op, just return success 108 if (inhdr.sample_rate == outhdr.sample_rate) { 109 return (AUDIO_SUCCESS); 110 } 111 112 // If nothing in the buffer, do the simple thing 113 if (length == 0.) { 114 inbuf->SetHeader(outhdr); 115 return (AUDIO_SUCCESS); 116 } 117 118 // Add some padding to the output buffer 119 i = 4 * ((input_rate / output_rate) + (output_rate / input_rate)); 120 length += outhdr.Samples_to_Time(i); 121 122 // Allocate a new buffer 123 outbuf = new AudioBuffer(length, "(SampleRate conversion buffer)"); 124 if (outbuf == 0) 125 return (AUDIO_UNIXERROR); 126 if (err = outbuf->SetHeader(outhdr)) { 127 delete outbuf; 128 return (err); 129 } 130 131 // here's where the guts go ... 132 nsamps = resampler.filter((short *)inbuf->GetAddress(), 133 (int)inbuf->GetHeader().Time_to_Samples(inbuf->GetLength()), 134 (short *)outbuf->GetAddress()); 135 136 // do a sanity check. did we write more bytes then we had 137 // available in the output buffer? 138 insamps = (unsigned int) 139 outbuf->GetHeader().Time_to_Samples(outbuf->GetSize()); 140 141 AUDIO_DEBUG((2, "TypeResample: after filter, insamps=%d, outsamps=%d\n", 142 insamps, nsamps)); 143 144 if (nsamps > outbuf->GetHeader().Time_to_Samples(outbuf->GetSize())) { 145 AudioStderrMsg(outbuf, AUDIO_NOERROR, Fatal, 146 (char *)"resample filter corrupted the heap"); 147 } 148 149 // set output size appropriately 150 outbuf->SetLength(outbuf->GetHeader().Samples_to_Time(nsamps)); 151 152 // This will delete the buffer 153 inbuf->Reference(); 154 inbuf->Dereference(); 155 156 inbuf = outbuf; 157 return (AUDIO_SUCCESS); 158 } 159 160 AudioError AudioTypeSampleRate:: 161 Flush( 162 AudioBuffer*& outbuf) 163 { 164 AudioHdr h; 165 Double pos; 166 int nsamp; 167 size_t cnt; 168 AudioError err; 169 unsigned char *tmpbuf; 170 171 if (outbuf == NULL) 172 return (AUDIO_SUCCESS); 173 h = outbuf->GetHeader(); 174 175 nsamp = resampler.getFlushSize(); 176 if (nsamp > 0) { 177 cnt = (size_t)nsamp * h.bytes_per_unit; 178 tmpbuf = new unsigned char[cnt]; 179 180 // this does a flush 181 nsamp = resampler.filter(NULL, 0, (short *)tmpbuf); 182 183 // Copy to the supplied buffer 184 if (nsamp > 0) { 185 cnt = (size_t)nsamp * h.bytes_per_unit; 186 pos = outbuf->GetLength(); 187 err = outbuf->AppendData(tmpbuf, cnt, pos); 188 if (err) 189 return (err); 190 } 191 delete tmpbuf; 192 } 193 return (AUDIO_SUCCESS); 194 } 195