git: f151b7fa349a - main - audio/webrtc-audio-processing: rebase c9bdb0ca7d0f on top of upstream
- Go to: [ bottom of page ] [ top of archives ] [ this month ]
Date: Fri, 02 Jun 2023 18:32:16 UTC
The branch main has been updated by jbeich:
URL: https://cgit.FreeBSD.org/ports/commit/?id=f151b7fa349aa9ba465a3526d1c7d569a422d728
commit f151b7fa349aa9ba465a3526d1c7d569a422d728
Author: Jan Beich <jbeich@FreeBSD.org>
AuthorDate: 2023-06-02 17:44:47 +0000
Commit: Jan Beich <jbeich@FreeBSD.org>
CommitDate: 2023-06-02 18:31:24 +0000
audio/webrtc-audio-processing: rebase c9bdb0ca7d0f on top of upstream
---
audio/webrtc-audio-processing/Makefile | 1 +
audio/webrtc-audio-processing/distinfo | 2 +
.../webrtc-audio-processing/files/patch-powerpc64 | 67 ++++++++++
.../files/patch-powerpc64.patch | 140 ---------------------
4 files changed, 70 insertions(+), 140 deletions(-)
diff --git a/audio/webrtc-audio-processing/Makefile b/audio/webrtc-audio-processing/Makefile
index 3ee52ce0cf06..ed81695cabdc 100644
--- a/audio/webrtc-audio-processing/Makefile
+++ b/audio/webrtc-audio-processing/Makefile
@@ -10,6 +10,7 @@ PATCHFILES+= 3f9907f93d39.patch:-p1 # https://gitlab.freedesktop.org/pulseaudio/
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
+PATCHFILES+= c18b3945a1cc.patch:-p1 # https://gitlab.freedesktop.org/pulseaudio/webrtc-audio-processing/-/merge_requests/17
MAINTAINER= jbeich@FreeBSD.org
COMMENT= AudioProcessing module from WebRTC project
diff --git a/audio/webrtc-audio-processing/distinfo b/audio/webrtc-audio-processing/distinfo
index 9458f9e257de..67c99850c4f7 100644
--- a/audio/webrtc-audio-processing/distinfo
+++ b/audio/webrtc-audio-processing/distinfo
@@ -11,3 +11,5 @@ SHA256 (d49a0855a33b.patch) = fe8a4421a664108e7f5223f61278cd5a9096d8d0f33d1d648a
SIZE (d49a0855a33b.patch) = 1388
SHA256 (f29ff57d6ccd.patch) = 9200b95da26ee34ef106cb3f2eed75d95eed7e9911a7632923b3d885409f9406
SIZE (f29ff57d6ccd.patch) = 858
+SHA256 (c18b3945a1cc.patch) = 3293a0f2a77e568efa1c1d6f6c845e9e31cd72491306c0b65e80db42bc261073
+SIZE (c18b3945a1cc.patch) = 5238
diff --git a/audio/webrtc-audio-processing/files/patch-powerpc64 b/audio/webrtc-audio-processing/files/patch-powerpc64
new file mode 100644
index 000000000000..b9982cf783a1
--- /dev/null
+++ b/audio/webrtc-audio-processing/files/patch-powerpc64
@@ -0,0 +1,67 @@
+Modified https://gitlab.freedesktop.org/pulseaudio/webrtc-audio-processing/-/merge_requests/17/diffs?commit_id=c18b3945a1cc for FreeBSD + pffft patch to include altivec.h.
+--- webrtc/common_audio/wav_file.cc.orig 2020-11-27 19:30:53 UTC
++++ webrtc/common_audio/wav_file.cc
+@@ -14,7 +14,7 @@
+
+ #include <algorithm>
+ #include <array>
+-#include <byteswap.h>
++#include <sys/endian.h>
+ #include <cstdio>
+ #include <type_traits>
+ #include <utility>
+@@ -123,7 +123,7 @@ size_t WavReader::ReadSamples(const size_t num_samples
+ }
+ #ifndef WEBRTC_ARCH_LITTLE_ENDIAN
+ for (size_t i = 0; i < num_samples; i++) {
+- samples[i] = bswap_16(samples[i]);
++ samples[i] = bswap16(samples[i]);
+ }
+ #endif
+
+@@ -171,7 +171,7 @@ size_t WavReader::ReadSamples(const size_t num_samples
+ #ifndef WEBRTC_ARCH_LITTLE_ENDIAN
+ // TODO: is this the right place for this?
+ for (size_t i = 0; i < num_samples; i++) {
+- samples[i] = bswap_16(samples[i]);
++ samples[i] = bswap16(samples[i]);
+ }
+ #endif
+
+@@ -229,7 +229,7 @@ void WavWriter::WriteSamples(const int16_t* samples, s
+ #else
+ std::array<int16_t, kMaxChunksize> converted_samples;
+ for (size_t j = 0; j < num_samples_to_write; ++j) {
+- converted_samples[j] = bswap_16(samples[i + j]);
++ converted_samples[j] = bswap16(samples[i + j]);
+ }
+ RTC_CHECK(
+ file_.Write(converted_samples.data(),
+@@ -241,7 +241,7 @@ void WavWriter::WriteSamples(const int16_t* samples, s
+ for (size_t j = 0; j < num_samples_to_write; ++j) {
+ int16_t sample = samples[i + j];
+ #ifndef WEBRTC_ARCH_LITTLE_ENDIAN
+- sample = bswap_16(sample);
++ sample = bswap16(sample);
+ #endif
+ converted_samples[j] = S16ToFloat(sample);
+ }
+@@ -267,7 +267,7 @@ void WavWriter::WriteSamples(const float* samples, siz
+ for (size_t j = 0; j < num_samples_to_write; ++j) {
+ int16_t sample = FloatS16ToS16(samples[i + j]);
+ #ifndef WEBRTC_ARCH_LITTLE_ENDIAN
+- sample = bswap_16(sample);
++ sample = bswap16(sample);
+ #endif
+ converted_samples[j] = sample;
+ }
+--- webrtc/third_party/pffft/src/pffft.c.orig 2020-11-27 19:30:53 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))
diff --git a/audio/webrtc-audio-processing/files/patch-powerpc64.patch b/audio/webrtc-audio-processing/files/patch-powerpc64.patch
deleted file mode 100644
index c7eba0e841cd..000000000000
--- a/audio/webrtc-audio-processing/files/patch-powerpc64.patch
+++ /dev/null
@@ -1,140 +0,0 @@
-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))