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 * Copyright 2019 RackTop Systems.
27 */
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
41AudioTypeSampleRate::
42AudioTypeSampleRate(int inrate, int outrate) :
43	resampler(inrate, outrate), input_rate(inrate), output_rate(outrate)
44{
45}
46
47// Destructor
48AudioTypeSampleRate::
49~AudioTypeSampleRate()
50{
51}
52
53// Test conversion possibilities.
54// Return TRUE if conversion to/from the specified type is possible.
55Boolean AudioTypeSampleRate::
56CanConvert(
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
73AudioError AudioTypeSampleRate::
74Convert(
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#ifdef DEBUG
84	size_t		insamps;
85#endif
86	AudioError	err;
87
88	inhdr = inbuf->GetHeader();
89	length = inbuf->GetLength();
90
91	if (Undefined(length)) {
92		return (AUDIO_ERR_BADARG);
93	}
94
95	// Make sure we're not being asked to do the impossible
96	// XXX - need a better error code
97	if ((err = inhdr.Validate()) || (err = outhdr.Validate())) {
98		return (err);
99	}
100
101	// If the requested conversion is different than what was initially
102	// established, then return an error.
103	// XXX - Maybe one day flush and re-init the filter
104	if ((inhdr.sample_rate != input_rate) ||
105	    (outhdr.sample_rate != output_rate)) {
106		return (AUDIO_ERR_BADARG);
107	}
108
109	// If conversion is a no-op, just return success
110	if (inhdr.sample_rate == outhdr.sample_rate) {
111		return (AUDIO_SUCCESS);
112	}
113
114	// If nothing in the buffer, do the simple thing
115	if (length == 0.) {
116		inbuf->SetHeader(outhdr);
117		return (AUDIO_SUCCESS);
118	}
119
120	// Add some padding to the output buffer
121	i = 4 * ((input_rate / output_rate) + (output_rate / input_rate));
122	length += outhdr.Samples_to_Time(i);
123
124	// Allocate a new buffer
125	outbuf = new AudioBuffer(length, "(SampleRate conversion buffer)");
126	if (outbuf == 0)
127		return (AUDIO_UNIXERROR);
128	if (err = outbuf->SetHeader(outhdr)) {
129		delete outbuf;
130		return (err);
131	}
132
133	// here's where the guts go ...
134	nsamps = resampler.filter((short *)inbuf->GetAddress(),
135		    (int)inbuf->GetHeader().Time_to_Samples(inbuf->GetLength()),
136		    (short *)outbuf->GetAddress());
137
138#ifdef DEBUG
139	// do a sanity check. did we write more bytes then we had
140	// available in the output buffer?
141	insamps = (unsigned int)
142		outbuf->GetHeader().Time_to_Samples(outbuf->GetSize());
143
144	AUDIO_DEBUG((2, "TypeResample: after filter, insamps=%d, outsamps=%d\n",
145		    insamps, nsamps));
146#endif
147
148	if (nsamps > outbuf->GetHeader().Time_to_Samples(outbuf->GetSize())) {
149		AudioStderrMsg(outbuf, AUDIO_NOERROR, Fatal,
150		    (char *)"resample filter corrupted the heap");
151	}
152
153	// set output size appropriately
154	outbuf->SetLength(outbuf->GetHeader().Samples_to_Time(nsamps));
155
156	// This will delete the buffer
157	inbuf->Reference();
158	inbuf->Dereference();
159
160	inbuf = outbuf;
161	return (AUDIO_SUCCESS);
162}
163
164AudioError AudioTypeSampleRate::
165Flush(
166	AudioBuffer*&	outbuf)
167{
168	AudioHdr	h;
169	Double		pos;
170	int		nsamp;
171	size_t		cnt;
172	AudioError	err;
173	unsigned char	*tmpbuf;
174
175	if (outbuf == NULL)
176		return (AUDIO_SUCCESS);
177	h = outbuf->GetHeader();
178
179	nsamp = resampler.getFlushSize();
180	if (nsamp > 0) {
181		cnt = (size_t)nsamp * h.bytes_per_unit;
182		tmpbuf = new unsigned char[cnt];
183
184		// this does a flush
185		nsamp = resampler.filter(NULL, 0, (short *)tmpbuf);
186
187		// Copy to the supplied buffer
188		if (nsamp > 0) {
189			cnt = (size_t)nsamp * h.bytes_per_unit;
190			pos = outbuf->GetLength();
191			err = outbuf->AppendData(tmpbuf, cnt, pos);
192			if (err)
193				return (err);
194		}
195		delete tmpbuf;
196	}
197	return (AUDIO_SUCCESS);
198}
199