[alsa-plugins-freeworld] Fix build

commit 2b88547bb354945d743af96344d73d96b47738c2 Author: leigh123linux <leigh123linux(a)googlemail.com&gt; Date: Sat Apr 14 13:32:15 2018 +0100 Fix build alsa-plugins-freeworld.spec | 8 +- ffmpeg35_buildfix.patch | 256 ++++++++++++++++++++++++++++++++++++++++++++ 2 files changed, 263 insertions(+), 1 deletion(-) --- diff --git a/alsa-plugins-freeworld.spec b/alsa-plugins-freeworld.spec index fe8b6ed..380ac91 100644 --- a/alsa-plugins-freeworld.spec +++ b/alsa-plugins-freeworld.spec @@ -1,12 +1,14 @@ Name: alsa-plugins-freeworld Version: 1.1.6 -Release: 2%{?dist} +Release: 3%{?dist} Summary: The ALSA Plugins - freeworld version # All packages are LGPLv2+ with the exception of samplerate which is GPLv2+ License: LGPLv2+ URL: http://www.alsa-project.org/ Source0: ftp://ftp.alsa-project.org/pub/plugins/alsa-plugins-%{version}.tar.bz2 Patch0: plugin-config.patch +# Based on https://patchwork.kernel.org/patch/9753853/ +Patch1: ffmpeg35_buildfix.patch BuildRequires: autoconf automake libtool BuildRequires: alsa-lib-devel >= 1.1.6 @@ -43,6 +45,7 @@ libavcodec's resampler. %prep %setup -q -n alsa-plugins-%{version}%{?prever} %patch0 -p1 -b .plugin-config +%patch1 -p1 -b .ffmpeg35_buildfix %build autoreconf -vif @@ -86,6 +89,9 @@ find $RPM_BUILD_ROOT -name "*.la" -exec rm {} \; %changelog +* Sat Apr 14 2018 Leigh Scott <leigh123linux(a)googlemail.com&gt; - 1.1.6-3 +- Fix build + * Fri Apr 13 2018 Jaroslav Kysela <perex(a)perex.cz&gt; - 1.1.6-2 - Use plugin config files from upstream, spec cleanups diff --git a/ffmpeg35_buildfix.patch b/ffmpeg35_buildfix.patch new file mode 100644 index 0000000..982794f --- /dev/null +++ b/ffmpeg35_buildfix.patch @@ -0,0 +1,256 @@ +--- a/rate-lavc/Makefile.am ++++ b/rate-lavc/Makefile.am +@@ -7,7 +7,7 @@ + AM_LDFLAGS = -module -avoid-version -export-dynamic -no-undefined $(LDFLAGS_NOUNDEFINED) + + libasound_module_rate_lavcrate_la_SOURCES = rate_lavcrate.c +-libasound_module_rate_lavcrate_la_LIBADD = @ALSA_LIBS@ @AVCODEC_LIBS@ ++libasound_module_rate_lavcrate_la_LIBADD = @ALSA_LIBS@ -lavresample -lavutil + + noinst_HEADERS = gcd.h + + +--- a/rate-lavc/rate_lavcrate.c ++++ b/rate-lavr/rate_lavcrate.c +@@ -1,9 +1,6 @@ + /* +- * Rate converter plugin using libavcodec's resampler +- * Copyright (c) 2007 by Nicholas Kain <njkain(a)gmail.com&gt; +- * +- * based on rate converter that uses libsamplerate +- * Copyright (c) 2006 by Takashi Iwai <tiwai(a)suse.de&gt; ++ * 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));