Attachment #398818 for bug #539680

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Lines 66-72 if test "$use_maemo_rm" = "yes"; then Link Here

(-)alsa-plugins-1.0.28/configure.ac (-2 / +18 lines)
66	fi	66	fi
67		67
68	AC_ARG_ENABLE([avcodec],	68	AC_ARG_ENABLE([avcodec],
69	AS_HELP_STRING([--disable-avcodec], [Don't build plugins depending on avcodec (a52)]))	69	AS_HELP_STRING([--disable-avcodec], [Do not build plugins depending on avcodec (a52)]))
70		70
71	if test "x$enable_avcodec" != "xno"; then	71	if test "x$enable_avcodec" != "xno"; then
72	PKG_CHECK_MODULES(AVCODEC, [libavcodec libavutil], [HAVE_AVCODEC=yes], [HAVE_AVCODEC=no])	72	PKG_CHECK_MODULES(AVCODEC, [libavcodec libavutil], [HAVE_AVCODEC=yes], [HAVE_AVCODEC=no])
Lines 101-106 if test $HAVE_AVCODEC = yes; then Link Here
101	if test -z "$AVCODEC_HEADER"; then	101	if test -z "$AVCODEC_HEADER"; then
102	HAVE_AVCODEC=no	102	HAVE_AVCODEC=no
103	fi	103	fi
		104	SAVE_LIBS=$LIBS
		105	LIBS="$LIBS $AVCODEC_LIBS"
		106	AC_CHECK_FUNCS([av_resample_init])
		107	LIBS=$SAVE_LIBS
104	fi	108	fi
105		109
106	AM_CONDITIONAL(HAVE_AVCODEC, test x$HAVE_AVCODEC = xyes)	110	AM_CONDITIONAL(HAVE_AVCODEC, test x$HAVE_AVCODEC = xyes)
Lines 108-113 AC_SUBST(AVCODEC_CFLAGS) Link Here
108	AC_SUBST(AVCODEC_LIBS)	112	AC_SUBST(AVCODEC_LIBS)
109	AC_SUBST(AVCODEC_HEADER)	113	AC_SUBST(AVCODEC_HEADER)
110		114
		115	AC_ARG_ENABLE([avresample],
		116	AS_HELP_STRING([--disable-avresample], [Do not build plugins depending on avcodec (lavrate)]))
		117
		118	if test "x$enable_avresample" != "xno"; then
		119	PKG_CHECK_MODULES(AVRESAMPLE, [libavresample libavutil], [HAVE_AVRESAMPLE=yes], [HAVE_AVRESAMPLE=no])
		120	fi
		121
		122	AM_CONDITIONAL(HAVE_AVRESAMPLE, test x$HAVE_AVCODEC = xyes)
		123	AC_SUBST(AVRESAMPLE_CFLAGS)
		124	AC_SUBST(AVRESAMPLE_LIBS)
		125	AC_SUBST(AVRESAMPLE_HEADER)
		126
111	PKG_CHECK_MODULES(speexdsp, [speexdsp >= 1.2], [HAVE_SPEEXDSP="yes"], [HAVE_SPEEXDSP=""])	127	PKG_CHECK_MODULES(speexdsp, [speexdsp >= 1.2], [HAVE_SPEEXDSP="yes"], [HAVE_SPEEXDSP=""])
112	AM_CONDITIONAL(HAVE_SPEEXDSP, test "$HAVE_SPEEXDSP" = "yes")	128	AM_CONDITIONAL(HAVE_SPEEXDSP, test "$HAVE_SPEEXDSP" = "yes")
113		129
Lines 181-187 AC_OUTPUT([ Link Here
181	mix/Makefile	197	mix/Makefile
182	rate/Makefile	198	rate/Makefile
183	a52/Makefile	199	a52/Makefile
184	rate-lavc/Makefile	200	rate-lavr/Makefile
185	maemo/Makefile	201	maemo/Makefile
186	doc/Makefile	202	doc/Makefile
187	usb_stream/Makefile	203	usb_stream/Makefile

Lines 9-16 if HAVE_SAMPLERATE Link Here

(-)alsa-plugins-1.0.28/Makefile.am (+6 lines)
9	SUBDIRS += rate	9	SUBDIRS += rate
10	endif	10	endif
11	if HAVE_AVCODEC	11	if HAVE_AVCODEC
		12	SUBDIRS += a52
		13	if !HAVE_AVRESAMPLE
12	SUBDIRS += a52 rate-lavc	14	SUBDIRS += a52 rate-lavc
13	endif	15	endif
		16	endif
		17	if HAVE_AVRESAMPLE
		18	SUBDIRS += rate-lavr
		19	endif
14	if HAVE_MAEMO_PLUGIN	20	if HAVE_MAEMO_PLUGIN
15	SUBDIRS += maemo	21	SUBDIRS += maemo
16	endif	22	endif




asound_module_rate_lavr_LTLIBRARIES = libasound_module_rate_lavr.la

asound_module_rate_lavrdir = @ALSA_PLUGIN_DIR@

AM_CFLAGS = -Wall -g @ALSA_CFLAGS@ @AVRESAMPLE_CFLAGS@
AM_LDFLAGS = -module -avoid-version -export-dynamic -no-undefined $(LDFLAGS_NOUNDEFINED)

libasound_module_rate_lavr_la_SOURCES = rate_lavr.c
libasound_module_rate_lavr_la_LIBADD = @ALSA_LIBS@ @AVRESAMPLE_LIBS@


install-exec-hook:
	rm -f $(DESTDIR)@ALSA_PLUGIN_DIR@/libasound_module_rate_lavcrate*.so
	$(LN_S) libasound_module_rate_lavr.so $(DESTDIR)@ALSA_PLUGIN_DIR@/libasound_module_rate_lavcrate.so
	$(LN_S) libasound_module_rate_lavr.so $(DESTDIR)@ALSA_PLUGIN_DIR@/libasound_module_rate_lavcrate_higher.so
	$(LN_S) libasound_module_rate_lavr.so $(DESTDIR)@ALSA_PLUGIN_DIR@/libasound_module_rate_lavcrate_high.so
	$(LN_S) libasound_module_rate_lavr.so $(DESTDIR)@ALSA_PLUGIN_DIR@/libasound_module_rate_lavcrate_fast.so
	$(LN_S) libasound_module_rate_lavr.so $(DESTDIR)@ALSA_PLUGIN_DIR@/libasound_module_rate_lavcrate_faster.so

uninstall-hook:
	rm -f $(DESTDIR)@ALSA_PLUGIN_DIR@/libasound_module_rate_lavcrate*.so
	rm -f $(DESTDIR)@ALSA_PLUGIN_DIR@/libasound_module_rate_lavr*.so




/*
 * Rate converter plugin using libavresample
 * Copyright (c) 2014 by Anton Khirnov
 *
 * 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; either
 * version 2.1 of the License, or (at your option) any later version.
 *
 * 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.
 */

#include <stdio.h>
#include <alsa/asoundlib.h>
#include <alsa/pcm_rate.h>

#include <libavresample/avresample.h>
#include <libavutil/channel_layout.h>
#include <libavutil/opt.h>
#include <libavutil/mathematics.h>
#include <libavutil/samplefmt.h>


static int filter_size = 16;
static int phase_shift = 10; /* auto-adjusts */
static double cutoff = 0; /* auto-adjusts */

struct rate_src {
	AVAudioResampleContext *avr;

	int in_rate;
	int out_rate;
	unsigned int channels;
};

static snd_pcm_uframes_t input_frames(void *obj, snd_pcm_uframes_t frames)
{
	return frames;
}

static snd_pcm_uframes_t output_frames(void *obj, snd_pcm_uframes_t frames)
{
	return frames;
}

static void pcm_src_free(void *obj)
{
	struct rate_src *rate = obj;
	avresample_free(&rate->avr);
}

static int pcm_src_init(void *obj, snd_pcm_rate_info_t *info)
{
	struct rate_src *rate = obj;
	int i, ir, or;

	if (!rate->avr || rate->channels != info->channels) {
		int ret;

		pcm_src_free(rate);
		rate->channels = info->channels;
		ir = rate->in_rate = info->in.rate;
		or = rate->out_rate = info->out.rate;
		i = av_gcd(or, ir);
		if (or > ir) {
			phase_shift = or/i;
		} else {
			phase_shift = ir/i;
		}
		if (cutoff <= 0.0) {
			cutoff = 1.0 - 1.0/filter_size;
			if (cutoff < 0.80)
				cutoff = 0.80;
		}

		rate->avr = avresample_alloc_context();
		if (!rate->avr)
			return -ENOMEM;

		av_opt_set_int(rate->avr, "in_sample_rate",     info->in.rate,  0);
		av_opt_set_int(rate->avr, "out_sample_rate",    info->out.rate, 0);
		av_opt_set_int(rate->avr, "in_sample_format",   AV_SAMPLE_FMT_S16, 0);
		av_opt_set_int(rate->avr, "out_sample_format",  AV_SAMPLE_FMT_S16, 0);
		av_opt_set_int(rate->avr, "in_channel_layout",  av_get_default_channel_layout(rate->channels), 0);
		av_opt_set_int(rate->avr, "out_channel_layout", av_get_default_channel_layout(rate->channels), 0);

		av_opt_set_int(rate->avr, "filter_size",        filter_size, 0);
		av_opt_set_int(rate->avr, "phase_shift",        phase_shift, 0);
		av_opt_set_double(rate->avr, "cutoff",          cutoff,      0);

		ret = avresample_open(rate->avr);
		if (ret < 0) {
			avresample_free(&rate->avr);
			return -EINVAL;
		}
	}

	return 0;
}

static int pcm_src_adjust_pitch(void *obj, snd_pcm_rate_info_t *info)
{
	struct rate_src *rate = obj;

	if (info->out.rate != rate->out_rate || info->in.rate != rate->in_rate)
		pcm_src_init(obj, info);
	return 0;
}

static void pcm_src_reset(void *obj)
{
	struct rate_src *rate = obj;

	if (rate->avr) {
		avresample_close(rate->avr);
		avresample_open(rate->avr);
	}
}

static void pcm_src_convert_s16(void *obj, int16_t *dst, unsigned int
	dst_frames, const int16_t *src, unsigned int src_frames)
{
	struct rate_src *rate = obj;
	int consumed = 0, chans=rate->channels, ret=0, i;
	int total_in = avresample_get_delay(rate->avr) + src_frames;

	ret = avresample_convert(rate->avr, &dst, dst_frames * chans * 2, dst_frames,
	                         &src, src_frames * chans * 2, src_frames);

	avresample_set_compensation(rate->avr,
                                    total_in - src_frames > filter_size ? 0 : 1, src_frames);
}

static void pcm_src_close(void *obj)
{
	pcm_src_free(obj);
}

#if SND_PCM_RATE_PLUGIN_VERSION >= 0x010002
static int get_supported_rates(void *obj, unsigned int *rate_min,
			       unsigned int *rate_max)
{
	*rate_min = *rate_max = 0; /* both unlimited */
	return 0;
}

static void dump(void *obj, snd_output_t *out)
{
	snd_output_printf(out, "Converter: libavr\n");
}
#endif

static snd_pcm_rate_ops_t pcm_src_ops = {
	.close = pcm_src_close,
	.init = pcm_src_init,
	.free = pcm_src_free,
	.adjust_pitch = pcm_src_adjust_pitch,
	.convert_s16 = pcm_src_convert_s16,
	.input_frames = input_frames,
	.output_frames = output_frames,
#if SND_PCM_RATE_PLUGIN_VERSION >= 0x010002
	.version = SND_PCM_RATE_PLUGIN_VERSION,
	.get_supported_rates = get_supported_rates,
	.dump = dump,
#endif
};

int pcm_src_open(unsigned int version, void **objp, snd_pcm_rate_ops_t *ops)

{
	struct rate_src *rate;

#if SND_PCM_RATE_PLUGIN_VERSION < 0x010002
	if (version != SND_PCM_RATE_PLUGIN_VERSION) {
		fprintf(stderr, "Invalid rate plugin version %x\n", version);
		return -EINVAL;
	}
#endif
	rate = calloc(1, sizeof(*rate));
	if (!rate)
		return -ENOMEM;

	*objp = rate;
	rate->avr = NULL;
#if SND_PCM_RATE_PLUGIN_VERSION >= 0x010002
	if (version == 0x010001)
		memcpy(ops, &pcm_src_ops, sizeof(snd_pcm_rate_old_ops_t));
	else
#endif
		*ops = pcm_src_ops;
	return 0;
}

int SND_PCM_RATE_PLUGIN_ENTRY(lavcrate)(unsigned int version, void **objp,
			snd_pcm_rate_ops_t *ops)
{
	return pcm_src_open(version, objp, ops);
}
int SND_PCM_RATE_PLUGIN_ENTRY(lavcrate_higher)(unsigned int version,
			void **objp, snd_pcm_rate_ops_t *ops)
{
	filter_size = 64;
	return pcm_src_open(version, objp, ops);
}
int SND_PCM_RATE_PLUGIN_ENTRY(lavcrate_high)(unsigned int version,
			void **objp, snd_pcm_rate_ops_t *ops)
{
	filter_size = 32;
	return pcm_src_open(version, objp, ops);
}
int SND_PCM_RATE_PLUGIN_ENTRY(lavcrate_fast)(unsigned int version,
			void **objp, snd_pcm_rate_ops_t *ops)
{
	filter_size = 8;
	return pcm_src_open(version, objp, ops);
}
int SND_PCM_RATE_PLUGIN_ENTRY(lavcrate_faster)(unsigned int version,
			void **objp, snd_pcm_rate_ops_t *ops)
{
	filter_size = 4;
	return pcm_src_open(version, objp, ops);
}



Return to bug 539680

Line 0 Link Here

(-)alsa-plugins-1.0.28/rate-lavr/Makefile.am (+22 lines)
	1	asound_module_rate_lavr_LTLIBRARIES = libasound_module_rate_lavr.la
	2
	3	asound_module_rate_lavrdir = @ALSA_PLUGIN_DIR@
	4
	5	AM_CFLAGS = -Wall -g @ALSA_CFLAGS@ @AVRESAMPLE_CFLAGS@
	6	AM_LDFLAGS = -module -avoid-version -export-dynamic -no-undefined $(LDFLAGS_NOUNDEFINED)
	7
	8	libasound_module_rate_lavr_la_SOURCES = rate_lavr.c
	9	libasound_module_rate_lavr_la_LIBADD = @ALSA_LIBS@ @AVRESAMPLE_LIBS@
	10
	11
	12	install-exec-hook:
	13	rm -f $(DESTDIR)@ALSA_PLUGIN_DIR@/libasound_module_rate_lavcrate*.so
	14	$(LN_S) libasound_module_rate_lavr.so $(DESTDIR)@ALSA_PLUGIN_DIR@/libasound_module_rate_lavcrate.so
	15	$(LN_S) libasound_module_rate_lavr.so $(DESTDIR)@ALSA_PLUGIN_DIR@/libasound_module_rate_lavcrate_higher.so
	16	$(LN_S) libasound_module_rate_lavr.so $(DESTDIR)@ALSA_PLUGIN_DIR@/libasound_module_rate_lavcrate_high.so
	17	$(LN_S) libasound_module_rate_lavr.so $(DESTDIR)@ALSA_PLUGIN_DIR@/libasound_module_rate_lavcrate_fast.so
	18	$(LN_S) libasound_module_rate_lavr.so $(DESTDIR)@ALSA_PLUGIN_DIR@/libasound_module_rate_lavcrate_faster.so
	19
	20	uninstall-hook:
	21	rm -f $(DESTDIR)@ALSA_PLUGIN_DIR@/libasound_module_rate_lavcrate*.so
	22	rm -f $(DESTDIR)@ALSA_PLUGIN_DIR@/libasound_module_rate_lavr*.so

Line 0 Link Here

(-)alsa-plugins-1.0.28/rate-lavr/rate_lavr.c (+227 lines)
		1	/*
		2	* Rate converter plugin using libavresample
		3	* Copyright (c) 2014 by Anton Khirnov
		4	*
		5	* This library is free software; you can redistribute it and/or
		6	* modify it under the terms of the GNU Lesser General Public
		7	* License as published by the Free Software Foundation; either
		8	* version 2.1 of the License, or (at your option) any later version.
		9	*
		10	* This library is distributed in the hope that it will be useful,
		11	* but WITHOUT ANY WARRANTY; without even the implied warranty of
		12	* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
		13	* Lesser General Public License for more details.
		14	*/
		15
		16	#include <stdio.h>
		17	#include <alsa/asoundlib.h>
		18	#include <alsa/pcm_rate.h>
		19
		20	#include <libavresample/avresample.h>
		21	#include <libavutil/channel_layout.h>
		22	#include <libavutil/opt.h>
		23	#include <libavutil/mathematics.h>
		24	#include <libavutil/samplefmt.h>
		25
		26
		27	static int filter_size = 16;
		28	static int phase_shift = 10; /* auto-adjusts */
		29	static double cutoff = 0; /* auto-adjusts */
		30
		31	struct rate_src {
		32	AVAudioResampleContext *avr;
		33
		34	int in_rate;
		35	int out_rate;
		36	unsigned int channels;
		37	};
		38
		39	static snd_pcm_uframes_t input_frames(void *obj, snd_pcm_uframes_t frames)
		40	{
		41	return frames;
		42	}
		43
		44	static snd_pcm_uframes_t output_frames(void *obj, snd_pcm_uframes_t frames)
		45	{
		46	return frames;
		47	}
		48
		49	static void pcm_src_free(void *obj)
		50	{
		51	struct rate_src *rate = obj;
		52	avresample_free(&rate->avr);
		53	}
		54
		55	static int pcm_src_init(void obj, snd_pcm_rate_info_t info)
		56	{
		57	struct rate_src *rate = obj;
		58	int i, ir, or;
		59
		60	if (!rate->avr \|\| rate->channels != info->channels) {
		61	int ret;
		62
		63	pcm_src_free(rate);
		64	rate->channels = info->channels;
65	ir = rate->in_rate = info->in.rate;
66	or = rate->out_rate = info->out.rate;
67	i = av_gcd(or, ir);
68	if (or > ir) {
69	phase_shift = or/i;
70	} else {
71	phase_shift = ir/i;
72	}
73	if (cutoff <= 0.0) {
74	cutoff = 1.0 - 1.0/filter_size;
75	if (cutoff < 0.80)
76	cutoff = 0.80;
77	}
78
79	rate->avr = avresample_alloc_context();
80	if (!rate->avr)
81	return -ENOMEM;
82
83	av_opt_set_int(rate->avr, "in_sample_rate", info->in.rate, 0);
84	av_opt_set_int(rate->avr, "out_sample_rate", info->out.rate, 0);
85	av_opt_set_int(rate->avr, "in_sample_format", AV_SAMPLE_FMT_S16, 0);
86	av_opt_set_int(rate->avr, "out_sample_format", AV_SAMPLE_FMT_S16, 0);
87	av_opt_set_int(rate->avr, "in_channel_layout", av_get_default_channel_layout(rate->channels), 0);
88	av_opt_set_int(rate->avr, "out_channel_layout", av_get_default_channel_layout(rate->channels), 0);
89
90	av_opt_set_int(rate->avr, "filter_size", filter_size, 0);
91	av_opt_set_int(rate->avr, "phase_shift", phase_shift, 0);
92	av_opt_set_double(rate->avr, "cutoff", cutoff, 0);
93
94	ret = avresample_open(rate->avr);
95	if (ret < 0) {
96	avresample_free(&rate->avr);
97	return -EINVAL;
98	}
99	}
100
101	return 0;
102	}
103
104	static int pcm_src_adjust_pitch(void obj, snd_pcm_rate_info_t info)
105	{
106	struct rate_src *rate = obj;
107
108	if (info->out.rate != rate->out_rate \|\| info->in.rate != rate->in_rate)
109	pcm_src_init(obj, info);
110	return 0;
111	}
112
113	static void pcm_src_reset(void *obj)
114	{
115	struct rate_src *rate = obj;
116
117	if (rate->avr) {
118	avresample_close(rate->avr);
119	avresample_open(rate->avr);
120	}
121	}
122
123	static void pcm_src_convert_s16(void obj, int16_t dst, unsigned int
124	dst_frames, const int16_t *src, unsigned int src_frames)
125	{
126	struct rate_src *rate = obj;
127	int consumed = 0, chans=rate->channels, ret=0, i;
128	int total_in = avresample_get_delay(rate->avr) + src_frames;
129
130	ret = avresample_convert(rate->avr, &dst, dst_frames * chans * 2, dst_frames,
131	&src, src_frames * chans * 2, src_frames);
132
133	avresample_set_compensation(rate->avr,
134	total_in - src_frames > filter_size ? 0 : 1, src_frames);
135	}
136
137	static void pcm_src_close(void *obj)
138	{
139	pcm_src_free(obj);
140	}
141
142	#if SND_PCM_RATE_PLUGIN_VERSION >= 0x010002
143	static int get_supported_rates(void obj, unsigned int rate_min,
144	unsigned int *rate_max)
145	{
146	rate_min = rate_max = 0; /* both unlimited */
147	return 0;
148	}
149
150	static void dump(void obj, snd_output_t out)
151	{
152	snd_output_printf(out, "Converter: libavr\n");
153	}
154	#endif
155
156	static snd_pcm_rate_ops_t pcm_src_ops = {
157	.close = pcm_src_close,
158	.init = pcm_src_init,
159	.free = pcm_src_free,
160	.adjust_pitch = pcm_src_adjust_pitch,
161	.convert_s16 = pcm_src_convert_s16,
162	.input_frames = input_frames,
163	.output_frames = output_frames,
164	#if SND_PCM_RATE_PLUGIN_VERSION >= 0x010002
165	.version = SND_PCM_RATE_PLUGIN_VERSION,
166	.get_supported_rates = get_supported_rates,
167	.dump = dump,
168	#endif
169	};
170
171	int pcm_src_open(unsigned int version, void *objp, snd_pcm_rate_ops_t ops)
172
173	{
174	struct rate_src *rate;
175
176	#if SND_PCM_RATE_PLUGIN_VERSION < 0x010002
177	if (version != SND_PCM_RATE_PLUGIN_VERSION) {
178	fprintf(stderr, "Invalid rate plugin version %x\n", version);
179	return -EINVAL;
180	}
181	#endif
182	rate = calloc(1, sizeof(*rate));
183	if (!rate)
184	return -ENOMEM;
185
186	*objp = rate;
187	rate->avr = NULL;
188	#if SND_PCM_RATE_PLUGIN_VERSION >= 0x010002
189	if (version == 0x010001)
190	memcpy(ops, &pcm_src_ops, sizeof(snd_pcm_rate_old_ops_t));
191	else
192	#endif
193	*ops = pcm_src_ops;
194	return 0;
195	}
196
197	int SND_PCM_RATE_PLUGIN_ENTRY(lavcrate)(unsigned int version, void **objp,
198	snd_pcm_rate_ops_t *ops)
199	{
200	return pcm_src_open(version, objp, ops);
201	}
202	int SND_PCM_RATE_PLUGIN_ENTRY(lavcrate_higher)(unsigned int version,
203	void *objp, snd_pcm_rate_ops_t ops)
204	{
205	filter_size = 64;
206	return pcm_src_open(version, objp, ops);
207	}
208	int SND_PCM_RATE_PLUGIN_ENTRY(lavcrate_high)(unsigned int version,
209	void *objp, snd_pcm_rate_ops_t ops)
210	{
211	filter_size = 32;
212	return pcm_src_open(version, objp, ops);
213	}
214	int SND_PCM_RATE_PLUGIN_ENTRY(lavcrate_fast)(unsigned int version,
215	void *objp, snd_pcm_rate_ops_t ops)
216	{
217	filter_size = 8;
218	return pcm_src_open(version, objp, ops);
219	}
220	int SND_PCM_RATE_PLUGIN_ENTRY(lavcrate_faster)(unsigned int version,
221	void *objp, snd_pcm_rate_ops_t ops)
222	{
223	filter_size = 4;
224	return pcm_src_open(version, objp, ops);
225	}
226
227