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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.
This commit is contained in:
Piotr Kubaj
parent
c7d08b1bc0
commit
c9bdb0ca7d
3 changed files with 147 additions and 2 deletions
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@ -8,6 +8,8 @@ PATCH_SITES= https://gitlab.freedesktop.org/pulseaudio/${PORTNAME}/-/commit/
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PATCHFILES+= 2083c9a5dd34.patch:-p1 # https://gitlab.freedesktop.org/pulseaudio/webrtc-audio-processing/-/merge_requests/6
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PATCHFILES+= 3f9907f93d39.patch:-p1 # https://gitlab.freedesktop.org/pulseaudio/webrtc-audio-processing/-/merge_requests/13
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PATCHFILES+= b34c1d5746ea.patch:-p1 # https://gitlab.freedesktop.org/pulseaudio/webrtc-audio-processing/-/merge_requests/14
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PATCHFILES+= d49a0855a33b.patch:-p1 # https://gitlab.freedesktop.org/pulseaudio/webrtc-audio-processing/-/merge_requests/17
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PATCHFILES+= f29ff57d6ccd.patch:-p1 # https://gitlab.freedesktop.org/pulseaudio/webrtc-audio-processing/-/merge_requests/17
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MAINTAINER= jbeich@FreeBSD.org
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COMMENT= AudioProcessing module from WebRTC project
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@ -16,7 +18,6 @@ LICENSE= BSD3CLAUSE
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LICENSE_FILE= ${WRKSRC}/COPYING
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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
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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
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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
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BUILD_DEPENDS= cmake:devel/cmake
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@ -1,4 +1,4 @@
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TIMESTAMP = 1606505453
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TIMESTAMP = 1645899872
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SHA256 (webrtc-audio-processing-1.0.tar.gz) = 441a30d2717b2eb4145c6eb96c2d5a270fe0b4bc71aebf76716750c47be1936f
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SIZE (webrtc-audio-processing-1.0.tar.gz) = 856721
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SHA256 (2083c9a5dd34.patch) = 3c34cc248c0292032b26ef67f078505037e51e84b6c7955162cac058aed54360
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@ -7,3 +7,7 @@ SHA256 (3f9907f93d39.patch) = 8e300c0ab9e85463ed0aebc7c69c95c293a14ef2ca0f50c7b7
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SIZE (3f9907f93d39.patch) = 2049
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SHA256 (b34c1d5746ea.patch) = b50edddb93a5bd4fdcc5110702111fe4ac000867358b7131e2ccce883a4c8dee
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SIZE (b34c1d5746ea.patch) = 923
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SHA256 (d49a0855a33b.patch) = fe8a4421a664108e7f5223f61278cd5a9096d8d0f33d1d648ac2c952d4c633b4
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SIZE (d49a0855a33b.patch) = 1388
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SHA256 (f29ff57d6ccd.patch) = 9200b95da26ee34ef106cb3f2eed75d95eed7e9911a7632923b3d885409f9406
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SIZE (f29ff57d6ccd.patch) = 858
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140
audio/webrtc-audio-processing/files/patch-powerpc64.patch
Normal file
140
audio/webrtc-audio-processing/files/patch-powerpc64.patch
Normal file
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@ -0,0 +1,140 @@
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Modified https://gitlab.freedesktop.org/pulseaudio/webrtc-audio-processing/-/merge_requests/17/diffs?commit_id=d49a0855a33bb56cc1935642c0d4bf7a3f474fbd for FreeBSD + pffft patch to include altivec.h.
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--- webrtc/common_audio/wav_file.cc
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+++ webrtc/common_audio/wav_file.cc
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@@ -14,6 +14,7 @@
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#include <algorithm>
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#include <array>
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+#include <sys/endian.h>
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#include <cstdio>
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#include <type_traits>
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#include <utility>
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@@ -89,10 +90,6 @@ void WavReader::Reset() {
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size_t WavReader::ReadSamples(const size_t num_samples,
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int16_t* const samples) {
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-#ifndef WEBRTC_ARCH_LITTLE_ENDIAN
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-#error "Need to convert samples to big-endian when reading from WAV file"
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-#endif
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-
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size_t num_samples_left_to_read = num_samples;
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size_t next_chunk_start = 0;
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while (num_samples_left_to_read > 0 && num_unread_samples_ > 0) {
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@@ -124,15 +121,16 @@ size_t WavReader::ReadSamples(const size_t num_samples,
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num_unread_samples_ -= num_samples_read;
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num_samples_left_to_read -= num_samples_read;
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}
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+#ifndef WEBRTC_ARCH_LITTLE_ENDIAN
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+ for (size_t i = 0; i < num_samples; i++) {
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+ samples[i] = bswap16(samples[i]);
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+ }
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+#endif
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return num_samples - num_samples_left_to_read;
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}
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size_t WavReader::ReadSamples(const size_t num_samples, float* const samples) {
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-#ifndef WEBRTC_ARCH_LITTLE_ENDIAN
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-#error "Need to convert samples to big-endian when reading from WAV file"
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-#endif
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-
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size_t num_samples_left_to_read = num_samples;
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size_t next_chunk_start = 0;
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while (num_samples_left_to_read > 0 && num_unread_samples_ > 0) {
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@@ -170,6 +168,12 @@ size_t WavReader::ReadSamples(const size_t num_samples, float* const samples) {
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num_unread_samples_ -= num_samples_read;
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num_samples_left_to_read -= num_samples_read;
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}
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+#ifndef WEBRTC_ARCH_LITTLE_ENDIAN
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+ // TODO: is this the right place for this?
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+ for (size_t i = 0; i < num_samples; i++) {
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+ samples[i] = bswap16(samples[i]);
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+ }
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+#endif
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return num_samples - num_samples_left_to_read;
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}
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@@ -213,23 +217,33 @@ WavWriter::WavWriter(FileWrapper file,
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}
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void WavWriter::WriteSamples(const int16_t* samples, size_t num_samples) {
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-#ifndef WEBRTC_ARCH_LITTLE_ENDIAN
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-#error "Need to convert samples to little-endian when writing to WAV file"
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-#endif
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-
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for (size_t i = 0; i < num_samples; i += kMaxChunksize) {
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const size_t num_remaining_samples = num_samples - i;
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const size_t num_samples_to_write =
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std::min(kMaxChunksize, num_remaining_samples);
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if (format_ == WavFormat::kWavFormatPcm) {
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+#ifndef WEBRTC_ARCH_LITTLE_ENDIAN
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RTC_CHECK(
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file_.Write(&samples[i], num_samples_to_write * sizeof(samples[0])));
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+#else
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+ std::array<int16_t, kMaxChunksize> converted_samples;
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+ for (size_t j = 0; j < num_samples_to_write; ++j) {
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+ converted_samples[j] = bswap16(samples[i + j]);
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+ }
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+ RTC_CHECK(
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+ file_.Write(converted_samples.data(),
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+ num_samples_to_write * sizeof(converted_samples[0])));
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+#endif
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} else {
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RTC_CHECK_EQ(format_, WavFormat::kWavFormatIeeeFloat);
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std::array<float, kMaxChunksize> converted_samples;
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for (size_t j = 0; j < num_samples_to_write; ++j) {
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- converted_samples[j] = S16ToFloat(samples[i + j]);
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+ int16_t sample = samples[i + j];
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+#ifndef WEBRTC_ARCH_LITTLE_ENDIAN
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+ sample = bswap16(sample);
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+#endif
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+ converted_samples[j] = S16ToFloat(sample);
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}
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RTC_CHECK(
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file_.Write(converted_samples.data(),
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@@ -243,10 +257,6 @@ void WavWriter::WriteSamples(const int16_t* samples, size_t num_samples) {
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}
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void WavWriter::WriteSamples(const float* samples, size_t num_samples) {
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-#ifndef WEBRTC_ARCH_LITTLE_ENDIAN
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-#error "Need to convert samples to little-endian when writing to WAV file"
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-#endif
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-
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for (size_t i = 0; i < num_samples; i += kMaxChunksize) {
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const size_t num_remaining_samples = num_samples - i;
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const size_t num_samples_to_write =
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@@ -255,7 +265,11 @@ void WavWriter::WriteSamples(const float* samples, size_t num_samples) {
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if (format_ == WavFormat::kWavFormatPcm) {
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std::array<int16_t, kMaxChunksize> converted_samples;
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for (size_t j = 0; j < num_samples_to_write; ++j) {
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- converted_samples[j] = FloatS16ToS16(samples[i + j]);
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+ int16_t sample = FloatS16ToS16(samples[i + j]);
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+#ifndef WEBRTC_ARCH_LITTLE_ENDIAN
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+ sample = bswap16(sample);
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+#endif
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+ converted_samples[j] = sample;
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}
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RTC_CHECK(
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file_.Write(converted_samples.data(),
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@@ -264,6 +278,7 @@ void WavWriter::WriteSamples(const float* samples, size_t num_samples) {
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RTC_CHECK_EQ(format_, WavFormat::kWavFormatIeeeFloat);
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std::array<float, kMaxChunksize> converted_samples;
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for (size_t j = 0; j < num_samples_to_write; ++j) {
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+ // TODO: is swap needed for big-endian here?
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converted_samples[j] = FloatS16ToFloat(samples[i + j]);
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}
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RTC_CHECK(
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--
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GitLab
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--- webrtc/third_party/pffft/src/pffft.c.orig 2022-02-26 18:37:29 UTC
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+++ webrtc/third_party/pffft/src/pffft.c
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@@ -100,6 +100,7 @@
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Altivec support macros
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*/
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#if !defined(PFFFT_SIMD_DISABLE) && (defined(__ppc__) || defined(__ppc64__))
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+#include <altivec.h>
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typedef vector float v4sf;
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# define SIMD_SZ 4
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# define VZERO() ((vector float) vec_splat_u8(0))
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