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::
AudioTypeSampleRate(int inrate,int outrate)42 AudioTypeSampleRate(int inrate, int outrate) :
43 resampler(inrate, outrate), input_rate(inrate), output_rate(outrate)
44 {
45 }
46
47 // Destructor
48 AudioTypeSampleRate::
~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::
CanConvert(AudioHdr h) const56 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::
Convert(AudioBuffer * & inbuf,AudioHdr outhdr)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::
Flush(AudioBuffer * & outbuf)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