Control: tags 971332 + patch Control: tags 971332 + pending Dear maintainer,
as I intend to upload ffmpeg without libavresample-dev soon, I've prepared an NMU for alsa-plugins (versioned as 1.2.2-2.1) and uploaded it to DELAYED/7. Please feel free to tell me if I should delay it longer. Cheers -- Sebastian Ramacher
diff -Nru alsa-plugins-1.2.2/debian/changelog alsa-plugins-1.2.2/debian/changelog
--- alsa-plugins-1.2.2/debian/changelog 2020-08-20 21:37:42.000000000 +0200
+++ alsa-plugins-1.2.2/debian/changelog 2021-08-15 20:18:32.000000000 +0200
@@ -1,3 +1,11 @@
+alsa-plugins (1.2.2-2.1) unstable; urgency=medium
+
+ * Non-maintainer upload.
+ * Apply upstream switch to support libswresample in favor of libavreample
+ (Closes: #971332)
+
+ -- Sebastian Ramacher <sramac...@debian.org> Sun, 15 Aug 2021 20:18:32 +0200
+
alsa-plugins (1.2.2-2) unstable; urgency=medium
[ Sebastien Bacher ]
diff -Nru alsa-plugins-1.2.2/debian/control alsa-plugins-1.2.2/debian/control
--- alsa-plugins-1.2.2/debian/control 2020-02-29 17:15:35.000000000 +0100
+++ alsa-plugins-1.2.2/debian/control 2021-08-15 20:18:32.000000000 +0200
@@ -8,13 +8,13 @@
Build-Depends: debhelper-compat (= 12),
libasound2-dev (>= 1.1.8),
libavcodec-dev,
- libavresample-dev,
libavutil-dev,
libdbus-1-dev (>= 1.4.12-3~),
libjack-dev (>= 1:0.121.0+svn4538-2~),
libpulse-dev (>= 0.99.1-1~),
libsamplerate0-dev | libsamplerate-dev,
- libspeexdsp-dev
+ libspeexdsp-dev,
+ libswresample-dev
Standards-Version: 4.5.0
Homepage: https://www.alsa-project.org/
Vcs-Git: https://salsa.debian.org/alsa-team/alsa-plugins.git
diff -Nru alsa-plugins-1.2.2/debian/patches/rate-lab-convert-tolibswresample.patch alsa-plugins-1.2.2/debian/patches/rate-lab-convert-tolibswresample.patch
--- alsa-plugins-1.2.2/debian/patches/rate-lab-convert-tolibswresample.patch 1970-01-01 01:00:00.000000000 +0100
+++ alsa-plugins-1.2.2/debian/patches/rate-lab-convert-tolibswresample.patch 2021-08-08 22:33:02.000000000 +0200
@@ -0,0 +1,201 @@
+From: Takashi Iwai <ti...@suse.de>
+Date: Wed, 16 Jun 2021 08:21:35 +0000 (+0200)
+Subject: rate-lav: Convert to libswresample
+X-Git-Url: https://git.alsa-project.org/?p=alsa-plugins.git;a=commitdiff_plain;h=8a3c0d795fbef5700c8cedcc82c6a337170c76ee
+
+rate-lav: Convert to libswresample
+
+The libavresample has been deprecated. Convert to the new API for
+libswresample.
+
+The phase shift and cutoff seem to be either redundant or not-working
+properly, so those are dropped.
+
+Signed-off-by: Takashi Iwai <ti...@suse.de>
+Signed-off-by: Jaroslav Kysela <pe...@perex.cz>
+---
+
+diff --git a/configure.ac b/configure.ac
+index d5fe529..860daa9 100644
+--- a/configure.ac
++++ b/configure.ac
+@@ -93,7 +93,7 @@ AC_ARG_ENABLE([libav],
+ AS_HELP_STRING([--disable-libav], [Do not build plugins depending on libav/ffmpeg (a52,lavrate...)]))
+
+ if test "x$enable_libav" != "xno"; then
+- PKG_CHECK_MODULES(LIBAV, [libavcodec libavutil libavresample], [HAVE_LIBAV=yes], [HAVE_LIBAV=no])
++ PKG_CHECK_MODULES(LIBAV, [libavcodec libavutil libswresample], [HAVE_LIBAV=yes], [HAVE_LIBAV=no])
+ fi
+
+ if test "x$HAVE_LIBAV" = "xno"; then
+diff --git a/doc/lavrate.txt b/doc/lavrate.txt
+index 6575183..fa6bbb0 100644
+--- a/doc/lavrate.txt
++++ b/doc/lavrate.txt
+@@ -2,7 +2,7 @@ Rate Converter Plugin Using libavresample
+ =========================================0
+
+ The plugin in rate-lavr subdirectory is an external rate converter using
+-libavresample library. You can use this rate converter plugin by defining a
++libswresample library. You can use this rate converter plugin by defining a
+ rate PCM with "converter" parameter, such as:
+
+ pcm.my_rate {
+@@ -16,7 +16,7 @@ The plug plugin has also a similar field, "rate_converter".
+ Or, more easily, define a global variable "defaults.pcm.rate_converter",
+ which is used as the default converter type by plug and rate plugins:
+
+- defaults.pcm.rate_converter "lavcrate"
++ defaults.pcm.rate_converter "lavrate"
+
+ Write the above in your ~/.asoundrc or /etc/asound.conf.
+
+diff --git a/rate-lav/rate_lavrate.c b/rate-lav/rate_lavrate.c
+index 2b992c5..e9c6740 100644
+--- a/rate-lav/rate_lavrate.c
++++ b/rate-lav/rate_lavrate.c
+@@ -1,5 +1,5 @@
+ /*
+- * Rate converter plugin using libavresample
++ * Rate converter plugin using libswresample
+ * Copyright (c) 2014 by Anton Khirnov
+ *
+ * This library is free software; you can redistribute it and/or
+@@ -17,7 +17,7 @@
+ #include <alsa/asoundlib.h>
+ #include <alsa/pcm_rate.h>
+
+-#include <libavresample/avresample.h>
++#include <libswresample/swresample.h>
+ #include <libavutil/channel_layout.h>
+ #include <libavutil/opt.h>
+ #include <libavutil/mathematics.h>
+@@ -25,11 +25,9 @@
+
+
+ static unsigned int filter_size = 16;
+-static unsigned int phase_shift = 10; /* auto-adjusts */
+-static double cutoff = 0; /* auto-adjusts */
+
+ struct rate_src {
+- AVAudioResampleContext *avr;
++ SwrContext *avr;
+
+ unsigned int in_rate;
+ unsigned int out_rate;
+@@ -51,51 +49,38 @@ static snd_pcm_uframes_t output_frames(void *obj ATTRIBUTE_UNUSED,
+ static void pcm_src_free(void *obj)
+ {
+ struct rate_src *rate = obj;
+- avresample_free(&rate->avr);
++ swr_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->in_rate = info->in.rate;
++ rate->out_rate = info->out.rate;
+
+- rate->avr = avresample_alloc_context();
++ rate->avr = swr_alloc();
+ 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_channel_layout(rate->avr, "in_channel_layout",
++ av_get_default_channel_layout(rate->channels), 0);
++ av_opt_set_channel_layout(rate->avr, "out_channel_layout",
++ av_get_default_channel_layout(rate->channels), 0);
++ av_opt_set_int(rate->avr, "in_sample_rate", rate->in_rate, 0);
++ av_opt_set_int(rate->avr, "out_sample_rate", rate->out_rate, 0);
++ av_opt_set_sample_fmt(rate->avr, "in_sample_fmt", AV_SAMPLE_FMT_S16, 0);
++ av_opt_set_sample_fmt(rate->avr, "out_sample_fmt", AV_SAMPLE_FMT_S16, 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);
++ ret = swr_init(rate->avr);
+ if (ret < 0) {
+- avresample_free(&rate->avr);
++ SNDERR("sw_init() error %d\n", ret);
++ swr_free(&rate->avr);
+ return -EINVAL;
+ }
+ }
+@@ -118,8 +103,8 @@ static void pcm_src_reset(void *obj)
+
+ if (rate->avr) {
+ #if 0
+- avresample_close(rate->avr);
+- avresample_open(rate->avr);
++ swr_free(rate->avr);
++ swr_init(rate->avr);
+ #endif
+ }
+ }
+@@ -130,14 +115,14 @@ static void pcm_src_convert_s16(void *obj, int16_t *dst,
+ unsigned int src_frames)
+ {
+ struct rate_src *rate = obj;
+- int chans = rate->channels;
+- unsigned int total_in = avresample_get_delay(rate->avr) + src_frames;
++ unsigned int total_in = swr_get_delay(rate->avr, rate->in_rate) + src_frames;
+
+- avresample_convert(rate->avr, (uint8_t **)&dst, dst_frames * chans * 2, dst_frames,
+- (uint8_t **)&src, src_frames * chans * 2, src_frames);
++ swr_convert(rate->avr, (uint8_t **)&dst, dst_frames,
++ (const uint8_t **)&src, src_frames);
+
+- avresample_set_compensation(rate->avr,
+- total_in - src_frames > filter_size ? 0 : 1, src_frames);
++ swr_set_compensation(rate->avr,
++ total_in - src_frames > filter_size ? 0 : 1,
++ src_frames);
+ }
+
+ static void pcm_src_close(void *obj)
+@@ -156,7 +141,7 @@ static int get_supported_rates(void *obj ATTRIBUTE_UNUSED,
+
+ static void dump(void *obj ATTRIBUTE_UNUSED, snd_output_t *out)
+ {
+- snd_output_printf(out, "Converter: libavr\n");
++ snd_output_printf(out, "Converter: libswresample\n");
+ }
+ #endif
+
+@@ -177,7 +162,6 @@ static snd_pcm_rate_ops_t pcm_src_ops = {
+ };
+
+ int pcm_src_open(unsigned int version, void **objp, snd_pcm_rate_ops_t *ops)
+-
+ {
+ struct rate_src *rate;
+
diff -Nru alsa-plugins-1.2.2/debian/patches/series alsa-plugins-1.2.2/debian/patches/series
--- alsa-plugins-1.2.2/debian/patches/series 2019-11-05 16:55:49.000000000 +0100
+++ alsa-plugins-1.2.2/debian/patches/series 2021-08-08 22:33:35.000000000 +0200
@@ -1 +1,2 @@
arcam-av_uses_pthreads.patch
+rate-lab-convert-tolibswresample.patch
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