audio/webrtc-audio-processing: fix build on powerpc64*

Two patches waiting for approval by upstream + one that needed to be
modified for FreeBSD (s/bswap_16/bswap16/) + making pffft include altivec.h.

audio/webrtc-audio-processing/Makefile

@@ -8,6 +8,8 @@ PATCH_SITES=	https://gitlab.freedesktop.org/pulseaudio/${PORTNAME}/-/commit/
 PATCHFILES+=	2083c9a5dd34.patch:-p1 # https://gitlab.freedesktop.org/pulseaudio/webrtc-audio-processing/-/merge_requests/6
 PATCHFILES+=	3f9907f93d39.patch:-p1 # https://gitlab.freedesktop.org/pulseaudio/webrtc-audio-processing/-/merge_requests/13
 PATCHFILES+=	b34c1d5746ea.patch:-p1 # https://gitlab.freedesktop.org/pulseaudio/webrtc-audio-processing/-/merge_requests/14
+PATCHFILES+=	d49a0855a33b.patch:-p1 # https://gitlab.freedesktop.org/pulseaudio/webrtc-audio-processing/-/merge_requests/17
+PATCHFILES+=	f29ff57d6ccd.patch:-p1 # https://gitlab.freedesktop.org/pulseaudio/webrtc-audio-processing/-/merge_requests/17
 
 MAINTAINER=	jbeich@FreeBSD.org
 COMMENT=	AudioProcessing module from WebRTC project
@@ -16,7 +18,6 @@ LICENSE=	BSD3CLAUSE
 LICENSE_FILE=	${WRKSRC}/COPYING
 
 BROKEN_powerpc=		fails to compile: ./webrtc/rtc_base/system/arch.h:54:2: Please add support for your architecture in rtc_base/system/arch.h
-BROKEN_powerpc64=	fails to compile: ./webrtc/rtc_base/system/arch.h:54:2: Please add support for your architecture in rtc_base/system/arch.h
 BROKEN_riscv64=		fails to compile: ./webrtc/rtc_base/system/arch.h:54:2: Please add support for your architecture in rtc_base/system/arch.h
 
 BUILD_DEPENDS=	cmake:devel/cmake

audio/webrtc-audio-processing/distinfo

@@ -1,4 +1,4 @@
-TIMESTAMP = 1606505453
+TIMESTAMP = 1645899872
 SHA256 (webrtc-audio-processing-1.0.tar.gz) = 441a30d2717b2eb4145c6eb96c2d5a270fe0b4bc71aebf76716750c47be1936f
 SIZE (webrtc-audio-processing-1.0.tar.gz) = 856721
 SHA256 (2083c9a5dd34.patch) = 3c34cc248c0292032b26ef67f078505037e51e84b6c7955162cac058aed54360
@@ -7,3 +7,7 @@ SHA256 (3f9907f93d39.patch) = 8e300c0ab9e85463ed0aebc7c69c95c293a14ef2ca0f50c7b7
 SIZE (3f9907f93d39.patch) = 2049
 SHA256 (b34c1d5746ea.patch) = b50edddb93a5bd4fdcc5110702111fe4ac000867358b7131e2ccce883a4c8dee
 SIZE (b34c1d5746ea.patch) = 923
+SHA256 (d49a0855a33b.patch) = fe8a4421a664108e7f5223f61278cd5a9096d8d0f33d1d648ac2c952d4c633b4
+SIZE (d49a0855a33b.patch) = 1388
+SHA256 (f29ff57d6ccd.patch) = 9200b95da26ee34ef106cb3f2eed75d95eed7e9911a7632923b3d885409f9406
+SIZE (f29ff57d6ccd.patch) = 858

audio/webrtc-audio-processing/files/patch-powerpc64.patch

@@ -0,0 +1,140 @@
+Modified https://gitlab.freedesktop.org/pulseaudio/webrtc-audio-processing/-/merge_requests/17/diffs?commit_id=d49a0855a33bb56cc1935642c0d4bf7a3f474fbd for FreeBSD + pffft patch to include altivec.h.
+--- webrtc/common_audio/wav_file.cc
++++ webrtc/common_audio/wav_file.cc
+@@ -14,6 +14,7 @@
+ 
+ #include <algorithm>
+ #include <array>
++#include <sys/endian.h>
+ #include <cstdio>
+ #include <type_traits>
+ #include <utility>
+@@ -89,10 +90,6 @@ void WavReader::Reset() {
+ 
+ size_t WavReader::ReadSamples(const size_t num_samples,
+                               int16_t* const samples) {
+-#ifndef WEBRTC_ARCH_LITTLE_ENDIAN
+-#error "Need to convert samples to big-endian when reading from WAV file"
+-#endif
+-
+   size_t num_samples_left_to_read = num_samples;
+   size_t next_chunk_start = 0;
+   while (num_samples_left_to_read > 0 && num_unread_samples_ > 0) {
+@@ -124,15 +121,16 @@ size_t WavReader::ReadSamples(const size_t num_samples,
+     num_unread_samples_ -= num_samples_read;
+     num_samples_left_to_read -= num_samples_read;
+   }
++#ifndef WEBRTC_ARCH_LITTLE_ENDIAN
++  for (size_t i = 0; i < num_samples; i++) {
++    samples[i] = bswap16(samples[i]);
++  }
++#endif
+ 
+   return num_samples - num_samples_left_to_read;
+ }
+ 
+ size_t WavReader::ReadSamples(const size_t num_samples, float* const samples) {
+-#ifndef WEBRTC_ARCH_LITTLE_ENDIAN
+-#error "Need to convert samples to big-endian when reading from WAV file"
+-#endif
+-
+   size_t num_samples_left_to_read = num_samples;
+   size_t next_chunk_start = 0;
+   while (num_samples_left_to_read > 0 && num_unread_samples_ > 0) {
+@@ -170,6 +168,12 @@ size_t WavReader::ReadSamples(const size_t num_samples, float* const samples) {
+     num_unread_samples_ -= num_samples_read;
+     num_samples_left_to_read -= num_samples_read;
+   }
++#ifndef WEBRTC_ARCH_LITTLE_ENDIAN
++  // TODO: is this the right place for this?
++  for (size_t i = 0; i < num_samples; i++) {
++    samples[i] = bswap16(samples[i]);
++  }
++#endif
+ 
+   return num_samples - num_samples_left_to_read;
+ }
+@@ -213,23 +217,33 @@ WavWriter::WavWriter(FileWrapper file,
+ }
+ 
+ void WavWriter::WriteSamples(const int16_t* samples, size_t num_samples) {
+-#ifndef WEBRTC_ARCH_LITTLE_ENDIAN
+-#error "Need to convert samples to little-endian when writing to WAV file"
+-#endif
+-
+   for (size_t i = 0; i < num_samples; i += kMaxChunksize) {
+     const size_t num_remaining_samples = num_samples - i;
+     const size_t num_samples_to_write =
+         std::min(kMaxChunksize, num_remaining_samples);
+ 
+     if (format_ == WavFormat::kWavFormatPcm) {
++#ifndef WEBRTC_ARCH_LITTLE_ENDIAN
+       RTC_CHECK(
+           file_.Write(&samples[i], num_samples_to_write * sizeof(samples[0])));
++#else
++      std::array<int16_t, kMaxChunksize> converted_samples;
++      for (size_t j = 0; j < num_samples_to_write; ++j) {
++        converted_samples[j] = bswap16(samples[i + j]);
++      }
++      RTC_CHECK(
++          file_.Write(converted_samples.data(),
++                      num_samples_to_write * sizeof(converted_samples[0])));
++#endif
+     } else {
+       RTC_CHECK_EQ(format_, WavFormat::kWavFormatIeeeFloat);
+       std::array<float, kMaxChunksize> converted_samples;
+       for (size_t j = 0; j < num_samples_to_write; ++j) {
+-        converted_samples[j] = S16ToFloat(samples[i + j]);
++        int16_t sample = samples[i + j];
++#ifndef WEBRTC_ARCH_LITTLE_ENDIAN
++        sample = bswap16(sample);
++#endif
++        converted_samples[j] = S16ToFloat(sample);
+       }
+       RTC_CHECK(
+           file_.Write(converted_samples.data(),
+@@ -243,10 +257,6 @@ void WavWriter::WriteSamples(const int16_t* samples, size_t num_samples) {
+ }
+ 
+ void WavWriter::WriteSamples(const float* samples, size_t num_samples) {
+-#ifndef WEBRTC_ARCH_LITTLE_ENDIAN
+-#error "Need to convert samples to little-endian when writing to WAV file"
+-#endif
+-
+   for (size_t i = 0; i < num_samples; i += kMaxChunksize) {
+     const size_t num_remaining_samples = num_samples - i;
+     const size_t num_samples_to_write =
+@@ -255,7 +265,11 @@ void WavWriter::WriteSamples(const float* samples, size_t num_samples) {
+     if (format_ == WavFormat::kWavFormatPcm) {
+       std::array<int16_t, kMaxChunksize> converted_samples;
+       for (size_t j = 0; j < num_samples_to_write; ++j) {
+-        converted_samples[j] = FloatS16ToS16(samples[i + j]);
++        int16_t sample = FloatS16ToS16(samples[i + j]);
++#ifndef WEBRTC_ARCH_LITTLE_ENDIAN
++        sample = bswap16(sample);
++#endif
++        converted_samples[j] = sample;
+       }
+       RTC_CHECK(
+           file_.Write(converted_samples.data(),
+@@ -264,6 +278,7 @@ void WavWriter::WriteSamples(const float* samples, size_t num_samples) {
+       RTC_CHECK_EQ(format_, WavFormat::kWavFormatIeeeFloat);
+       std::array<float, kMaxChunksize> converted_samples;
+       for (size_t j = 0; j < num_samples_to_write; ++j) {
++        // TODO: is swap needed for big-endian here?
+         converted_samples[j] = FloatS16ToFloat(samples[i + j]);
+       }
+       RTC_CHECK(
+-- 
+GitLab
+
+--- webrtc/third_party/pffft/src/pffft.c.orig	2022-02-26 18:37:29 UTC
++++ webrtc/third_party/pffft/src/pffft.c
+@@ -100,6 +100,7 @@
+    Altivec support macros 
+ */
+ #if !defined(PFFFT_SIMD_DISABLE) && (defined(__ppc__) || defined(__ppc64__))
++#include <altivec.h>
+ typedef vector float v4sf;
+ #  define SIMD_SZ 4
+ #  define VZERO() ((vector float) vec_splat_u8(0))