May 2017 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH] ALSA: Adjust structure(snd_timer_tread) members to avoid 8 padding bytes
by Manjeet Pawar 30 May '17

30 May '17

This patch is made to fix below CVE issue where 8 padding bytes are sent to user without being initialized. Structure members are adjusted to avoid 8 padding bytes. CVE-2016-4569/CVE-2016-4578: The stack object tread has a total size of 32 bytes. Its field event and val both contain 4 bytes padding. These 8 bytes padding bytes are sent to user without being initialized. Fix patch: https://git.kernel.org/pub/scm/linux/kernel/git/torvalds/linux.git/commit/?… https://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git/commit/?h=f… These patches does memset on structure object to initialize 8 padding bytes, which can be skipped by adjusting structure members. Signed-off-by: Manjeet Pawar <manjeet.p(a)samsung.com> Signed-off-by: Vaneet Narang <v.narang(a)samsung.com> --- include/uapi/sound/asound.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/include/uapi/sound/asound.h b/include/uapi/sound/asound.h index a45be6b..820c5a0 100644 --- a/include/uapi/sound/asound.h +++ b/include/uapi/sound/asound.h @@ -789,8 +789,8 @@ enum { struct snd_timer_tread { int event; - struct timespec tstamp; unsigned int val; + struct timespec tstamp; }; /**************************************************************************** -- 1.9.1

2 1

[alsa-devel] [PATCH - Fix building alsa-plugins against libav-10 1/1] Fix building alsa-plugins against libav-10
by plevine457＠gmail.com 30 May '17

30 May '17

From: Peter Levine <plevine457(a)gmail.com> Signed-off-by: Peter Levine <plevine457(a)gmail.com> diff --git a/Makefile.am b/Makefile.am index 69cfe0d..9195b56 100644 --- a/Makefile.am +++ b/Makefile.am @@ -9,8 +9,14 @@ if HAVE_SAMPLERATE SUBDIRS += rate endif if HAVE_AVCODEC +SUBDIRS += a52 +if !HAVE_AVRESAMPLE SUBDIRS += a52 rate-lavc endif +endif +if HAVE_AVRESAMPLE +SUBDIRS += rate-lavr +endif if HAVE_MAEMO_PLUGIN SUBDIRS += maemo endif diff --git a/configure.ac b/configure.ac index f42601c..0af5ec9 100644 --- a/configure.ac +++ b/configure.ac @@ -127,6 +127,10 @@ if test $HAVE_AVCODEC = yes; then if test -z "$AVCODEC_HEADER"; then HAVE_AVCODEC=no fi + SAVE_LIBS=$LIBS + LIBS="$LIBS $AVCODEC_LIBS" + AC_CHECK_FUNCS([av_resample_init]) + LIBS=$SAVE_LIBS CFLAGS="$CFLAGS_saved" fi @@ -135,6 +139,18 @@ AC_SUBST(AVCODEC_CFLAGS) AC_SUBST(AVCODEC_LIBS) AC_SUBST(AVCODEC_HEADER) +AC_ARG_ENABLE([avresample], + AS_HELP_STRING([--disable-avresample], [Do not build plugins depending on avcodec (lavrate)])) + +if test "x$enable_avresample" != "xno"; then + PKG_CHECK_MODULES(AVRESAMPLE, [libavresample libavutil], [HAVE_AVRESAMPLE=yes], [HAVE_AVRESAMPLE=no]) +fi + +AM_CONDITIONAL(HAVE_AVRESAMPLE, test x$HAVE_AVCODEC = xyes) +AC_SUBST(AVRESAMPLE_CFLAGS) +AC_SUBST(AVRESAMPLE_LIBS) +AC_SUBST(AVRESAMPLE_HEADER) + AC_ARG_ENABLE([speexdsp], AS_HELP_STRING([--disable-speexdsp], [Disable building of speexdsp plugin])) @@ -217,7 +233,7 @@ AC_OUTPUT([ mix/Makefile rate/Makefile a52/Makefile - rate-lavc/Makefile + rate-lavr/Makefile maemo/Makefile doc/Makefile usb_stream/Makefile diff --git a/rate-lavr/Makefile.am b/rate-lavr/Makefile.am new file mode 100644 index 0000000..a1dca35 --- /dev/null +++ b/rate-lavr/Makefile.am @@ -0,0 +1,22 @@ +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 diff --git a/rate-lavc/rate_lavcrate.c b/rate-lavr/rate_lavr.c similarity index 58% copy from rate-lavc/rate_lavcrate.c copy to rate-lavr/rate_lavr.c index 14a2198..fe3bf4b 100644 --- a/rate-lavc/rate_lavcrate.c +++ b/rate-lavr/rate_lavr.c @@ -1,9 +1,6 @@ /* - * Rate converter plugin using libavcodec's resampler - * Copyright (c) 2007 by Nicholas Kain <njkain(a)gmail.com> - * - * based on rate converter that uses libsamplerate - * Copyright (c) 2006 by Takashi Iwai <tiwai(a)suse.de> + * 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 @@ -19,21 +16,23 @@ #include <stdio.h> #include <alsa/asoundlib.h> #include <alsa/pcm_rate.h> -#include AVCODEC_HEADER -#include "gcd.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 { - struct AVResampleContext *context; + AVAudioResampleContext *avr; + int in_rate; int out_rate; - int stored; - int point; - int16_t **out; - int16_t **in; unsigned int channels; }; @@ -50,26 +49,7 @@ static snd_pcm_uframes_t output_frames(void *obj, snd_pcm_uframes_t frames) static void pcm_src_free(void *obj) { struct rate_src *rate = obj; - int i; - - if (rate->out) { - for (i=0; i<rate->channels; i++) { - free(rate->out[i]); - } - free(rate->out); - } - if (rate->in) { - for (i=0; i<rate->channels; i++) { - free(rate->in[i]); - } - free(rate->in); - } - rate->out = rate->in = NULL; - - if (rate->context) { - av_resample_close(rate->context); - rate->context = NULL; - } + avresample_free(&rate->avr); } static int pcm_src_init(void *obj, snd_pcm_rate_info_t *info) @@ -77,12 +57,14 @@ static int pcm_src_init(void *obj, snd_pcm_rate_info_t *info) struct rate_src *rate = obj; int i, ir, or; - if (! rate->context || rate->channels != info->channels) { + 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 = gcd(or, ir); + i = av_gcd(or, ir); if (or > ir) { phase_shift = or/i; } else { @@ -93,25 +75,27 @@ static int pcm_src_init(void *obj, snd_pcm_rate_info_t *info) if (cutoff < 0.80) cutoff = 0.80; } - rate->context = av_resample_init(info->out.rate, info->in.rate, - filter_size, phase_shift, - (info->out.rate >= info->in.rate ? 0 : 1), cutoff); - if (!rate->context) - return -EINVAL; - } - rate->out = malloc(rate->channels * sizeof(int16_t *)); - rate->in = malloc(rate->channels * sizeof(int16_t *)); - for (i=0; i<rate->channels; i++) { - rate->out[i] = calloc(info->out.period_size * 2, - sizeof(int16_t)); - rate->in[i] = calloc(info->in.period_size * 2, - sizeof(int16_t)); - } - rate->point = info->in.period_size / 2; - if (!rate->out || !rate->in) { - pcm_src_free(rate); - return -ENOMEM; + 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; @@ -129,48 +113,10 @@ static int pcm_src_adjust_pitch(void *obj, snd_pcm_rate_info_t *info) static void pcm_src_reset(void *obj) { struct rate_src *rate = obj; - rate->stored = 0; -} -static void deinterleave(const int16_t *src, int16_t **dst, unsigned int frames, - unsigned int chans, int overflow) -{ - int i, j; - - if (chans == 1) { - memcpy(dst + overflow, src, frames*sizeof(int16_t)); - } else if (chans == 2) { - for (j=overflow; j<(frames + overflow); j++) { - dst[0][j] = *(src++); - dst[1][j] = *(src++); - } - } else { - for (j=overflow; j<(frames + overflow); j++) { - for (i=0; i<chans; i++) { - dst[i][j] = *(src++); - } - } - } -} - -static void reinterleave(int16_t **src, int16_t *dst, unsigned int frames, - unsigned int chans) -{ - int i, j; - - if (chans == 1) { - memcpy(dst, src, frames*sizeof(int16_t)); - } else if (chans == 2) { - for (j=0; j<frames; j++) { - *(dst++) = src[0][j]; - *(dst++) = src[1][j]; - } - } else { - for (j=0; j<frames; j++) { - for (i=0; i<chans; i++) { - *(dst++) = src[i][j]; - } - } + if (rate->avr) { + avresample_close(rate->avr); + avresample_open(rate->avr); } } @@ -179,22 +125,13 @@ static void pcm_src_convert_s16(void *obj, int16_t *dst, unsigned int { struct rate_src *rate = obj; int consumed = 0, chans=rate->channels, ret=0, i; - int total_in = rate->stored + src_frames, new_stored; - - deinterleave(src, rate->in, src_frames, chans, rate->point); - for (i=0; i<chans; ++i) { - ret = av_resample(rate->context, rate->out[i], - rate->in[i]+rate->point-rate->stored, &consumed, - total_in, dst_frames, i == (chans - 1)); - new_stored = total_in-consumed; - memmove(rate->in[i]+rate->point-new_stored, - rate->in[i]+rate->point-rate->stored+consumed, - new_stored*sizeof(int16_t)); - } - av_resample_compensate(rate->context, - total_in-src_frames>filter_size?0:1, src_frames); - reinterleave(rate->out, dst, ret, chans); - rate->stored = total_in-consumed; + 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) @@ -212,7 +149,7 @@ static int get_supported_rates(void *obj, unsigned int *rate_min, static void dump(void *obj, snd_output_t *out) { - snd_output_printf(out, "Converter: liblavc\n"); + snd_output_printf(out, "Converter: libavr\n"); } #endif @@ -220,7 +157,6 @@ static snd_pcm_rate_ops_t pcm_src_ops = { .close = pcm_src_close, .init = pcm_src_init, .free = pcm_src_free, - .reset = pcm_src_reset, .adjust_pitch = pcm_src_adjust_pitch, .convert_s16 = pcm_src_convert_s16, .input_frames = input_frames, @@ -248,7 +184,7 @@ int pcm_src_open(unsigned int version, void **objp, snd_pcm_rate_ops_t *ops) return -ENOMEM; *objp = rate; - rate->context = NULL; + 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)); -- 2.13.0

2 1

[alsa-devel] [PATCH - Fix building alsa-plugins against libav-10 1/1] Fix building alsa-plugins against libav-10
by plevine457＠gmail.com 30 May '17

30 May '17

From: Peter Levine <plevine457(a)gmail.com> Signed-off-by: Peter Levine <plevine457(a)gmail.com> diff --git a/Makefile.am b/Makefile.am index 69cfe0d..9195b56 100644 --- a/Makefile.am +++ b/Makefile.am @@ -9,8 +9,14 @@ if HAVE_SAMPLERATE SUBDIRS += rate endif if HAVE_AVCODEC +SUBDIRS += a52 +if !HAVE_AVRESAMPLE SUBDIRS += a52 rate-lavc endif +endif +if HAVE_AVRESAMPLE +SUBDIRS += rate-lavr +endif if HAVE_MAEMO_PLUGIN SUBDIRS += maemo endif diff --git a/configure.ac b/configure.ac index f42601c..0af5ec9 100644 --- a/configure.ac +++ b/configure.ac @@ -127,6 +127,10 @@ if test $HAVE_AVCODEC = yes; then if test -z "$AVCODEC_HEADER"; then HAVE_AVCODEC=no fi + SAVE_LIBS=$LIBS + LIBS="$LIBS $AVCODEC_LIBS" + AC_CHECK_FUNCS([av_resample_init]) + LIBS=$SAVE_LIBS CFLAGS="$CFLAGS_saved" fi @@ -135,6 +139,18 @@ AC_SUBST(AVCODEC_CFLAGS) AC_SUBST(AVCODEC_LIBS) AC_SUBST(AVCODEC_HEADER) +AC_ARG_ENABLE([avresample], + AS_HELP_STRING([--disable-avresample], [Do not build plugins depending on avcodec (lavrate)])) + +if test "x$enable_avresample" != "xno"; then + PKG_CHECK_MODULES(AVRESAMPLE, [libavresample libavutil], [HAVE_AVRESAMPLE=yes], [HAVE_AVRESAMPLE=no]) +fi + +AM_CONDITIONAL(HAVE_AVRESAMPLE, test x$HAVE_AVCODEC = xyes) +AC_SUBST(AVRESAMPLE_CFLAGS) +AC_SUBST(AVRESAMPLE_LIBS) +AC_SUBST(AVRESAMPLE_HEADER) + AC_ARG_ENABLE([speexdsp], AS_HELP_STRING([--disable-speexdsp], [Disable building of speexdsp plugin])) @@ -217,7 +233,7 @@ AC_OUTPUT([ mix/Makefile rate/Makefile a52/Makefile - rate-lavc/Makefile + rate-lavr/Makefile maemo/Makefile doc/Makefile usb_stream/Makefile diff --git a/rate-lavr/Makefile.am b/rate-lavr/Makefile.am new file mode 100644 index 0000000..a1dca35 --- /dev/null +++ b/rate-lavr/Makefile.am @@ -0,0 +1,22 @@ +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 diff --git a/rate-lavr/rate_lavr.c b/rate-lavr/rate_lavr.c new file mode 100644 index 0000000..fe3bf4b --- /dev/null +++ b/rate-lavr/rate_lavr.c @@ -0,0 +1,227 @@ +/* + * 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); +} + + -- 2.13.0

2 1

[alsa-devel] [PATCH] ASoC: zx-i2s: fix harsh noise with 16-bit audio
by Shawn Guo 29 May '17

29 May '17

From: Shawn Guo <shawn.guo(a)linaro.org> The audio parameter setup in zx_i2s_hw_params() works fine with 32-bit samples, but for 16-bit ones, all we can hear is harsh noises. As suggested by vendor driver code, DMA burst size and TS width should always be 32 bits, no matter audio sample is 16-bit or 32-bit. Follow the suggestion above to fix the harsh noise issue seen with 16-bit audio samples. Signed-off-by: Shawn Guo <shawn.guo(a)linaro.org> --- sound/soc/zte/zx-i2s.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) diff --git a/sound/soc/zte/zx-i2s.c b/sound/soc/zte/zx-i2s.c index a865f37c2a56..a7f7a56e0a2d 100644 --- a/sound/soc/zte/zx-i2s.c +++ b/sound/soc/zte/zx-i2s.c @@ -226,11 +226,12 @@ static int zx_i2s_hw_params(struct snd_pcm_substream *substream, struct zx_i2s_info *i2s = snd_soc_dai_get_drvdata(socdai); struct snd_dmaengine_dai_dma_data *dma_data; unsigned int lane, ch_num, len, ret = 0; + unsigned int ts_width = 32; unsigned long val; unsigned long chn_cfg; dma_data = snd_soc_dai_get_dma_data(socdai, substream); - dma_data->addr_width = params_width(params) >> 3; + dma_data->addr_width = ts_width >> 3; val = readl_relaxed(i2s->reg_base + ZX_I2S_TIMING_CTRL); val &= ~(ZX_I2S_TIMING_TS_WIDTH_MASK | ZX_I2S_TIMING_DATA_SIZE_MASK | @@ -251,7 +252,7 @@ static int zx_i2s_hw_params(struct snd_pcm_substream *substream, dev_err(socdai->dev, "Unknown data format\n"); return -EINVAL; } - val |= ZX_I2S_TIMING_TS_WIDTH(len) | ZX_I2S_TIMING_DATA_SIZE(len); + val |= ZX_I2S_TIMING_TS_WIDTH(ts_width) | ZX_I2S_TIMING_DATA_SIZE(len); ch_num = params_channels(params); switch (ch_num) { -- 1.9.1

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[alsa-devel] [PATCH 0/3] ASoC: add audio-graph-scu-card
by Kuninori Morimoto 29 May '17

29 May '17

Hi Mark These adds audio-graph-scu-card which is same as simple-scu-card + OF-graph. 1) patch cleanups existing simple-scu-card document Kuninori Morimoto (3): 1) ASoC: simple-scu-card: cleanup documentation 2) ASoC: add audio-graph-scu-card document 3) ASoC: add audio-graph-scu-card support .../bindings/sound/audio-graph-scu-card.txt | 72 ++++ .../devicetree/bindings/sound/simple-scu-card.txt | 65 ++-- sound/soc/generic/Kconfig | 9 + sound/soc/generic/Makefile | 2 + sound/soc/generic/audio-graph-scu-card.c | 416 +++++++++++++++++++++ 5 files changed, 523 insertions(+), 41 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/audio-graph-scu-card.txt create mode 100644 sound/soc/generic/audio-graph-scu-card.c -- 1.9.1

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[alsa-devel] [PATCH 0/4] ASoC: tidyup return method from probe()
by Kuninori Morimoto 29 May '17

29 May '17

Hi Mark As Dan reported, current simple / simple-scu / audio-graph / audio-grach-scu(it is now posting) are using strange return method on probe() These patches fix these up. Kuninori Morimoto (4): ASoC: simple-card: tidyup return method from probe() ASoC: simple-scu-card: tidyup return method from probe() ASoC: audio-graph-card: tidyup return method from probe() ASoC: audio-graph-scu-card: tidyup return method from probe() sound/soc/generic/audio-graph-card.c | 6 ++++-- sound/soc/generic/audio-graph-scu-card.c | 6 ++++-- sound/soc/generic/simple-card.c | 6 ++++-- sound/soc/generic/simple-scu-card.c | 6 ++++-- 4 files changed, 16 insertions(+), 8 deletions(-) -- 1.9.1

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[alsa-devel] [PATCH 00/14] Revised patchset for PCM in-kernel copy support
by Takashi Iwai 26 May '17

26 May '17

Hi, this is a sort of RFC, a revised patchset of the previous two patches for killing set_fs() usages in PCM. Instead of converting to the merged copy_silence ops, this adds a new copy_kernel ops instead. At the same time, copy/silence are changed to receive the position and size in bytes instead of frames. This allows us to simplify the PCM core code. As a result, a good amount of code could be removed from pcm_lib.c. Note that this patchset isn't complete, containing only two driver conversions (nm256 for interleaved and rme9652 for non-interleaved) as a demonstration. Takashi === Takashi Iwai (14): ALSA: pcm: Introduce copy_user, copy_kernel and fill_silence ops ALSA: nm256: Convert to new PCM copy ops ALSA: rme9652: Convert to the new PCM ops ALSA: pcm: Drop the old copy and silence ops ALSA: pcm: Check PCM state by a common helper function ALSA: pcm: Shuffle codes ALSA: pcm: Call directly the common read/write helpers ALSA: pcm: More unification of PCM transfer codes ALSA: pcm: Unify read/write loop ALSA: pcm: Simplify snd_pcm_playback_silence() ALSA: pcm: Direct in-kernel read/write support usb: gadget: u_uac1: Kill set_fs() usage ALSA: pcm: Kill set_fs() in PCM OSS layer ALSA: pcm: Build OSS writev/readv helpers conditionally drivers/usb/gadget/function/u_uac1.c | 7 +- include/sound/pcm.h | 80 ++++- sound/core/oss/io.c | 4 +- sound/core/oss/pcm_oss.c | 81 +---- sound/core/oss/pcm_plugin.h | 6 +- sound/core/pcm_lib.c | 551 +++++++++++++---------------------- sound/pci/nm256/nm256.c | 57 ++-- sound/pci/rme9652/rme9652.c | 71 +++-- sound/soc/soc-pcm.c | 5 +- 9 files changed, 386 insertions(+), 476 deletions(-) -- 2.13.0

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[alsa-devel] Steinberg MIDEX8 driver attempt
by Hedde Bosman 26 May '17

26 May '17

Dear Alsa developers, I've owned a MIDEX8 for some years now and wanted to do something with it in Linux , so I started some analysis and coding. The current result can be found at https://github.com/sgorpi/midex8 it looks a bit like the standard midi class driver, but it requires some periodic (i guess timer) messages on endpoint 0x82. Therefore, I've took quite some inspiration from the sound/usb/midi.c driver. In general, it seems to work now for simple cases, but I'm having a few issues, that might be due to my lack of knowledge on the USB subsystem, or something else. In the (simplified) libusb test in the above repository, I do not seem to have these issues. Issue 1: Not all (interrupt) urbs that are submitted will complete (within 25 ms at least). Wireshark doesn't show the packets being sent out, and the completeion handler never gets called for those urbs, but also no errors are given by usb_submit_urb. This happens mainly with the timer messages, and my current solution is to unlink the urbs that haven't completed within 25ms. However, I feel there is something going wrong there. If I do not send urbs on the midi-in endpoint, I do not see any missing urbs... Do I have to mutex anything for a shared urb queue or so? Issue 2: Sometimes after sending a few midi messages, an empty urb shows up in wireshark. However, in code, there's an if-statement that should allow only messages of length > 0 to be submitted (line 568, if (num_read > 0)). Might this have to do with issue 1? I intend to improve on the driver some more, but if you could give me any advice on the above issues that would be highly appreciated. With kind regards, Hedde

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[alsa-devel] [PATCH 0/2] ALSA: pcm: add local header for module-local functions
by Takashi Sakamoto 26 May '17

26 May '17

Hi, This patchset is a supplement for merged patchset[1]. As I suggested[2], an added helper function is useful to some parts of PCM core module. Although it's better to replace existent code block with it, there's an issue to share it with several objects, because there is no header files for external symbols not exported to kernel space. This patchset adds a header file for the module-local symbols, then adds calls of the function in several parts of the module. [1] [alsa-devel] [PATCH 0/7] ALSA: Fix/improve PCM ack callback http://mailman.alsa-project.org/pipermail/alsa-devel/2017-May/120886.html [2] [alsa-devel] [PATCH 0/7] ALSA: Fix/improve PCM ack callback http://mailman.alsa-project.org/pipermail/alsa-devel/2017-May/120910.html Takashi Sakamoto (2): ALSA: pcm: add local header file for snd-pcm module ALSA: pcm: add arrangement for applying appl_ptr include/sound/pcm.h | 32 ----------------------------- sound/core/pcm.c | 2 ++ sound/core/pcm_lib.c | 14 +++++++------ sound/core/pcm_local.h | 54 +++++++++++++++++++++++++++++++++++++++++++++++++ sound/core/pcm_misc.c | 3 +++ sound/core/pcm_native.c | 13 +++++++----- 6 files changed, 75 insertions(+), 43 deletions(-) create mode 100644 sound/core/pcm_local.h -- 2.11.0

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[alsa-devel] [PATCH] pcm: dmix: Workaround for binary incompatibility
by Takashi Iwai 26 May '17

26 May '17

The commit 1a9bd0f04481 ("pcm: direct: Fix for sync issue on xrun recover") introduced a new field "recoveries" in snd_pcm_direct_share_t. Unfortunately this caused two issues: - It changed the size of the struct which is used as the magic key - The struct size differs between 32bit and 64bit due to alignment The former brought the incompatibility with the older alsa-lib, e.g. when you run an app with an older alsa-lib via LD_PRELOAD, it doesn't work any longer. The latter is more serious, it disallows running 32bit apps dmix with 64bit together. As a workaround, put recoveries field to the unused field "s.xfer_align", so that the struct is in an old form. This is a one-time workaround, and we may need to reconsider more about a breakage in future... Reported-and-tested-by: Cheng Sun <chengsun9(a)gmail.com> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- src/pcm/pcm_direct.c | 8 ++++---- src/pcm/pcm_direct.h | 3 +-- 2 files changed, 5 insertions(+), 6 deletions(-) diff --git a/src/pcm/pcm_direct.c b/src/pcm/pcm_direct.c index 89468d3537d0..43702601b0f2 100644 --- a/src/pcm/pcm_direct.c +++ b/src/pcm/pcm_direct.c @@ -615,7 +615,7 @@ int snd_pcm_direct_slave_recover(snd_pcm_direct_t *direct) } return ret; } - direct->shmptr->recoveries++; + direct->shmptr->s.recoveries++; semerr = snd_pcm_direct_semaphore_up(direct, DIRECT_IPC_SEM_CLIENT); if (semerr < 0) { @@ -631,11 +631,11 @@ int snd_pcm_direct_slave_recover(snd_pcm_direct_t *direct) */ int snd_pcm_direct_client_chk_xrun(snd_pcm_direct_t *direct, snd_pcm_t *pcm) { - if (direct->shmptr->recoveries != direct->recoveries) { + if (direct->shmptr->s.recoveries != direct->recoveries) { /* no matter how many xruns we missed - * so don't increment but just update to actual counter */ - direct->recoveries = direct->shmptr->recoveries; + direct->recoveries = direct->shmptr->s.recoveries; pcm->fast_ops->drop(pcm); /* trigger_tstamp update is missing in drop callbacks */ gettimestamp(&direct->trigger_tstamp, pcm->tstamp_type); @@ -1539,7 +1539,7 @@ int snd_pcm_direct_open_secondary_client(snd_pcm_t **spcmp, snd_pcm_direct_t *dm dmix->slave_buffer_size = spcm->buffer_size; dmix->slave_period_size = dmix->shmptr->s.period_size; dmix->slave_boundary = spcm->boundary; - dmix->recoveries = dmix->shmptr->recoveries; + dmix->recoveries = dmix->shmptr->s.recoveries; ret = snd_pcm_mmap(spcm); if (ret < 0) { diff --git a/src/pcm/pcm_direct.h b/src/pcm/pcm_direct.h index 24e85f096310..582f055a5451 100644 --- a/src/pcm/pcm_direct.h +++ b/src/pcm/pcm_direct.h @@ -66,7 +66,6 @@ typedef struct { char socket_name[256]; /* name of communication socket */ snd_pcm_type_t type; /* PCM type (currently only hw) */ int use_server; - unsigned int recoveries; /* no of executed recoveries on slave*/ struct { unsigned int format; snd_interval_t rate; @@ -95,7 +94,7 @@ typedef struct { unsigned int stop_threshold; unsigned int silence_threshold; unsigned int silence_size; - unsigned int xfer_align; /* not used */ + unsigned int recoveries; /* no of executed recoveries on slave*/ unsigned long long boundary; unsigned int info; unsigned int msbits; -- 2.13.0

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