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gnu: webrtc-audio-processing: Fix build.
* gnu/packages/audio.scm (webrtc-audio-processing) [source]: Drop patch and snippet. [native-inputs]: Add pkg-config. * gnu/packages/patches/webrtc-audio-processing-big-endian.patch: Delete file. * gnu/local.mk (dist_patch_DATA): De-register it. Change-Id: I3340371a8d484a0ad1faddedc911169e29957281
This commit is contained in:
Maxim Cournoyer
Ludovic Courtès
parent
1cf91238a2
commit
0ed04bbfe2
3 changed files with 2 additions and 359 deletions
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@ -2274,7 +2274,6 @@ dist_patch_DATA = \
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%D%/packages/patches/wcstools-extend-makefiles.patch \
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%D%/packages/patches/wdl-link-libs-and-fix-jnetlib.patch \
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%D%/packages/patches/webkitgtk-adjust-bubblewrap-paths.patch \
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%D%/packages/patches/webrtc-audio-processing-big-endian.patch \
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%D%/packages/patches/webrtc-for-telegram-desktop-unbundle-libsrtp.patch \
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%D%/packages/patches/websocketpp-fix-for-cmake-3.15.patch \
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%D%/packages/patches/wlroots-hwdata-fallback.patch \
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@ -313,34 +313,9 @@ (define-public webrtc-audio-processing
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(string-append "http://freedesktop.org/software/pulseaudio/"
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name "/" name "-" version ".tar.gz"))
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(sha256
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(base32 "0xfvq5lxg612vfzk3zk6896zcb4cgrrb7fq76w9h40magz0jymcm"))
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(modules '((guix build utils)))
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(snippet
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#~(begin
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;; See:
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;; <https://gitlab.freedesktop.org/pulseaudio/webrtc-audio-processing/-/issues/4>.
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(substitute* "meson.build"
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(("absl_flags_registry") "absl_flags_reflection"))
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(substitute* "webrtc/rtc_base/system/arch.h"
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(("defined\\(__aarch64__\\)" all)
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(string-append
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;; powerpc-linux
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"(defined(__PPC__) && __SIZEOF_SIZE_T__ == 4)\n"
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"#define WEBRTC_ARCH_32_BITS\n"
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"#define WEBRTC_ARCH_BIG_ENDIAN\n"
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;; powerpc64-linux
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"#elif (defined(__PPC64__) && defined(_BIG_ENDIAN))\n"
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"#define WEBRTC_ARCH_64_BITS\n"
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"#define WEBRTC_ARCH_BIG_ENDIAN\n"
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;; aarch64-linux
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"#elif " all
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;; riscv64-linux
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" || (defined(__riscv) && __riscv_xlen == 64)"
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;; powerpc64le-linux
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" || (defined(__PPC64__) && defined(_LITTLE_ENDIAN))")))))
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(patches
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(search-patches "webrtc-audio-processing-big-endian.patch"))))
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(base32 "0xfvq5lxg612vfzk3zk6896zcb4cgrrb7fq76w9h40magz0jymcm"))))
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(build-system meson-build-system)
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(native-inputs (list pkg-config))
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(inputs (list abseil-cpp))
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(synopsis "WebRTC's Audio Processing Library")
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(description "WebRTC-Audio-Processing library based on Google's
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@ -1,331 +0,0 @@
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https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/issues/127
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https://github.com/desktop-app/tg_owt/commit/65f002e
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From 65f002eeda1d97ddc70c8c49ec563987203c76f5 Mon Sep 17 00:00:00 2001
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From: Nicholas Guriev <nicholas@guriev.su>
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Date: Thu, 28 Jan 2021 20:54:06 +0300
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Subject: [PATCH] Provide endianness converters before writing or after reading
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WAV
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---
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src/common_audio/wav_file.cc | 80 ++++++++++++++++++++++++++-------
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src/common_audio/wav_header.cc | 81 ++++++++++++++++++++--------------
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2 files changed, 111 insertions(+), 50 deletions(-)
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diff --git a/src/common_audio/wav_file.cc b/src/common_audio/wav_file.cc
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index e49126f1..b5292668 100644
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--- a/webrtc/common_audio/wav_file.cc
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+++ b/webrtc/common_audio/wav_file.cc
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@@ -10,6 +10,7 @@
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#include "common_audio/wav_file.h"
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+#include <byteswap.h>
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#include <errno.h>
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#include <algorithm>
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@@ -34,6 +35,38 @@ bool FormatSupported(WavFormat format) {
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format == WavFormat::kWavFormatIeeeFloat;
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}
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+template <typename T>
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+void TranslateEndianness(T* destination, const T* source, size_t length) {
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+ static_assert(sizeof(T) == 2 || sizeof(T) == 4 || sizeof(T) == 8,
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+ "no converter, use integral types");
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+ if (sizeof(T) == 2) {
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+ const uint16_t* src = reinterpret_cast<const uint16_t*>(source);
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+ uint16_t* dst = reinterpret_cast<uint16_t*>(destination);
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+ for (size_t index = 0; index < length; index++) {
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+ dst[index] = bswap_16(src[index]);
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+ }
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+ }
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+ if (sizeof(T) == 4) {
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+ const uint32_t* src = reinterpret_cast<const uint32_t*>(source);
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+ uint32_t* dst = reinterpret_cast<uint32_t*>(destination);
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+ for (size_t index = 0; index < length; index++) {
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+ dst[index] = bswap_32(src[index]);
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+ }
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+ }
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+ if (sizeof(T) == 8) {
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+ const uint64_t* src = reinterpret_cast<const uint64_t*>(source);
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+ uint64_t* dst = reinterpret_cast<uint64_t*>(destination);
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+ for (size_t index = 0; index < length; index++) {
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+ dst[index] = bswap_64(src[index]);
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+ }
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+ }
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+}
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+
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+template <typename T>
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+void TranslateEndianness(T* buffer, size_t length) {
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+ TranslateEndianness(buffer, buffer, length);
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+}
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+
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// Doesn't take ownership of the file handle and won't close it.
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class WavHeaderFileReader : public WavHeaderReader {
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public:
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@@ -89,10 +122,6 @@ 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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@@ -105,6 +134,9 @@ size_t WavReader::ReadSamples(const size_t num_samples,
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num_bytes_read = file_.Read(samples_to_convert.data(),
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chunk_size * sizeof(samples_to_convert[0]));
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num_samples_read = num_bytes_read / sizeof(samples_to_convert[0]);
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+#ifdef WEBRTC_ARCH_BIG_ENDIAN
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+ TranslateEndianness(samples_to_convert.data(), num_samples_read);
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+#endif
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for (size_t j = 0; j < num_samples_read; ++j) {
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samples[next_chunk_start + j] = FloatToS16(samples_to_convert[j]);
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@@ -114,6 +146,10 @@ size_t WavReader::ReadSamples(const size_t num_samples,
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num_bytes_read = file_.Read(&samples[next_chunk_start],
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chunk_size * sizeof(samples[0]));
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num_samples_read = num_bytes_read / sizeof(samples[0]);
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+
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+#ifdef WEBRTC_ARCH_BIG_ENDIAN
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+ TranslateEndianness(&samples[next_chunk_start], num_samples_read);
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+#endif
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}
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RTC_CHECK(num_samples_read == 0 || (num_bytes_read % num_samples_read) == 0)
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<< "Corrupt file: file ended in the middle of a sample.";
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@@ -129,10 +165,6 @@ size_t WavReader::ReadSamples(const size_t num_samples,
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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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@@ -145,6 +177,9 @@ size_t WavReader::ReadSamples(const size_t num_samples, float* const samples) {
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num_bytes_read = file_.Read(samples_to_convert.data(),
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chunk_size * sizeof(samples_to_convert[0]));
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num_samples_read = num_bytes_read / sizeof(samples_to_convert[0]);
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+#ifdef WEBRTC_ARCH_BIG_ENDIAN
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+ TranslateEndianness(samples_to_convert.data(), num_samples_read);
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+#endif
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for (size_t j = 0; j < num_samples_read; ++j) {
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samples[next_chunk_start + j] =
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@@ -155,6 +190,9 @@ size_t WavReader::ReadSamples(const size_t num_samples, float* const samples) {
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num_bytes_read = file_.Read(&samples[next_chunk_start],
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chunk_size * sizeof(samples[0]));
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num_samples_read = num_bytes_read / sizeof(samples[0]);
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+#ifdef WEBRTC_ARCH_BIG_ENDIAN
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+ TranslateEndianness(&samples[next_chunk_start], num_samples_read);
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+#endif
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for (size_t j = 0; j < num_samples_read; ++j) {
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samples[next_chunk_start + j] =
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@@ -213,24 +251,32 @@ 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_BIG_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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+ TranslateEndianness(converted_samples.data(), &samples[i],
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+ num_samples_to_write);
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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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}
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+#ifdef WEBRTC_ARCH_BIG_ENDIAN
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+ TranslateEndianness(converted_samples.data(), num_samples_to_write);
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+#endif
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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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@@ -243,10 +289,6 @@ 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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@@ -257,6 +299,9 @@ void WavWriter::WriteSamples(const float* samples, size_t num_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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}
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+#ifdef WEBRTC_ARCH_BIG_ENDIAN
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+ TranslateEndianness(converted_samples.data(), num_samples_to_write);
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+#endif
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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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@@ -266,6 +311,9 @@ void WavWriter::WriteSamples(const float* samples, size_t num_samples) {
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for (size_t j = 0; j < num_samples_to_write; ++j) {
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converted_samples[j] = FloatS16ToFloat(samples[i + j]);
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}
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+#ifdef WEBRTC_ARCH_BIG_ENDIAN
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+ TranslateEndianness(converted_samples.data(), num_samples_to_write);
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+#endif
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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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diff --git a/webrtc/common_audio/wav_header.cc b/webrtc/common_audio/wav_header.cc
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index 1ccbffca..98264a5c 100644
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--- a/src/common_audio/wav_header.cc
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+++ b/src/common_audio/wav_header.cc
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@@ -14,6 +14,8 @@
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#include "common_audio/wav_header.h"
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+#include <endian.h>
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+
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#include <cstring>
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#include <limits>
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#include <string>
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@@ -26,10 +28,6 @@
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namespace webrtc {
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namespace {
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-#ifndef WEBRTC_ARCH_LITTLE_ENDIAN
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-#error "Code not working properly for big endian platforms."
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-#endif
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-
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#pragma pack(2)
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struct ChunkHeader {
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uint32_t ID;
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@@ -174,6 +172,8 @@ bool FindWaveChunk(ChunkHeader* chunk_header,
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if (readable->Read(chunk_header, sizeof(*chunk_header)) !=
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sizeof(*chunk_header))
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return false; // EOF.
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+ chunk_header->Size = le32toh(chunk_header->Size);
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+
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if (ReadFourCC(chunk_header->ID) == sought_chunk_id)
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return true; // Sought chunk found.
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// Ignore current chunk by skipping its payload.
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@@ -187,6 +187,13 @@ bool ReadFmtChunkData(FmtPcmSubchunk* fmt_subchunk, WavHeaderReader* readable) {
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if (readable->Read(&(fmt_subchunk->AudioFormat), kFmtPcmSubchunkSize) !=
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kFmtPcmSubchunkSize)
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return false;
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+ fmt_subchunk->AudioFormat = le16toh(fmt_subchunk->AudioFormat);
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+ fmt_subchunk->NumChannels = le16toh(fmt_subchunk->NumChannels);
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+ fmt_subchunk->SampleRate = le32toh(fmt_subchunk->SampleRate);
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+ fmt_subchunk->ByteRate = le32toh(fmt_subchunk->ByteRate);
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+ fmt_subchunk->BlockAlign = le16toh(fmt_subchunk->BlockAlign);
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+ fmt_subchunk->BitsPerSample = le16toh(fmt_subchunk->BitsPerSample);
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+
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const uint32_t fmt_size = fmt_subchunk->header.Size;
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if (fmt_size != kFmtPcmSubchunkSize) {
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// There is an optional two-byte extension field permitted to be present
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@@ -214,19 +221,22 @@ void WritePcmWavHeader(size_t num_channels,
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auto header = rtc::MsanUninitialized<WavHeaderPcm>({});
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const size_t bytes_in_payload = bytes_per_sample * num_samples;
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- header.riff.header.ID = PackFourCC('R', 'I', 'F', 'F');
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- header.riff.header.Size = RiffChunkSize(bytes_in_payload, *header_size);
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- header.riff.Format = PackFourCC('W', 'A', 'V', 'E');
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- header.fmt.header.ID = PackFourCC('f', 'm', 't', ' ');
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- header.fmt.header.Size = kFmtPcmSubchunkSize;
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- header.fmt.AudioFormat = MapWavFormatToHeaderField(WavFormat::kWavFormatPcm);
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- header.fmt.NumChannels = static_cast<uint16_t>(num_channels);
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- header.fmt.SampleRate = sample_rate;
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- header.fmt.ByteRate = ByteRate(num_channels, sample_rate, bytes_per_sample);
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- header.fmt.BlockAlign = BlockAlign(num_channels, bytes_per_sample);
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- header.fmt.BitsPerSample = static_cast<uint16_t>(8 * bytes_per_sample);
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- header.data.header.ID = PackFourCC('d', 'a', 't', 'a');
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- header.data.header.Size = static_cast<uint32_t>(bytes_in_payload);
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+ header.riff.header.ID = htole32(PackFourCC('R', 'I', 'F', 'F'));
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+ header.riff.header.Size =
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+ htole32(RiffChunkSize(bytes_in_payload, *header_size));
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+ header.riff.Format = htole32(PackFourCC('W', 'A', 'V', 'E'));
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+ header.fmt.header.ID = htole32(PackFourCC('f', 'm', 't', ' '));
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+ header.fmt.header.Size = htole32(kFmtPcmSubchunkSize);
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+ header.fmt.AudioFormat =
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+ htole16(MapWavFormatToHeaderField(WavFormat::kWavFormatPcm));
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+ header.fmt.NumChannels = htole16(num_channels);
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+ header.fmt.SampleRate = htole32(sample_rate);
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+ header.fmt.ByteRate =
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+ htole32(ByteRate(num_channels, sample_rate, bytes_per_sample));
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+ header.fmt.BlockAlign = htole16(BlockAlign(num_channels, bytes_per_sample));
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+ header.fmt.BitsPerSample = htole16(8 * bytes_per_sample);
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+ header.data.header.ID = htole32(PackFourCC('d', 'a', 't', 'a'));
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+ header.data.header.Size = htole32(bytes_in_payload);
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// Do an extra copy rather than writing everything to buf directly, since buf
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// might not be correctly aligned.
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@@ -245,24 +255,26 @@ void WriteIeeeFloatWavHeader(size_t num_channels,
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auto header = rtc::MsanUninitialized<WavHeaderIeeeFloat>({});
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const size_t bytes_in_payload = bytes_per_sample * num_samples;
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- header.riff.header.ID = PackFourCC('R', 'I', 'F', 'F');
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- header.riff.header.Size = RiffChunkSize(bytes_in_payload, *header_size);
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- header.riff.Format = PackFourCC('W', 'A', 'V', 'E');
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- header.fmt.header.ID = PackFourCC('f', 'm', 't', ' ');
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- header.fmt.header.Size = kFmtIeeeFloatSubchunkSize;
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+ header.riff.header.ID = htole32(PackFourCC('R', 'I', 'F', 'F'));
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+ header.riff.header.Size =
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+ htole32(RiffChunkSize(bytes_in_payload, *header_size));
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+ header.riff.Format = htole32(PackFourCC('W', 'A', 'V', 'E'));
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+ header.fmt.header.ID = htole32(PackFourCC('f', 'm', 't', ' '));
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+ header.fmt.header.Size = htole32(kFmtIeeeFloatSubchunkSize);
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header.fmt.AudioFormat =
|
||||
- MapWavFormatToHeaderField(WavFormat::kWavFormatIeeeFloat);
|
||||
- header.fmt.NumChannels = static_cast<uint16_t>(num_channels);
|
||||
- header.fmt.SampleRate = sample_rate;
|
||||
- header.fmt.ByteRate = ByteRate(num_channels, sample_rate, bytes_per_sample);
|
||||
- header.fmt.BlockAlign = BlockAlign(num_channels, bytes_per_sample);
|
||||
- header.fmt.BitsPerSample = static_cast<uint16_t>(8 * bytes_per_sample);
|
||||
- header.fmt.ExtensionSize = 0;
|
||||
- header.fact.header.ID = PackFourCC('f', 'a', 'c', 't');
|
||||
- header.fact.header.Size = 4;
|
||||
- header.fact.SampleLength = static_cast<uint32_t>(num_channels * num_samples);
|
||||
- header.data.header.ID = PackFourCC('d', 'a', 't', 'a');
|
||||
- header.data.header.Size = static_cast<uint32_t>(bytes_in_payload);
|
||||
+ htole16(MapWavFormatToHeaderField(WavFormat::kWavFormatIeeeFloat));
|
||||
+ header.fmt.NumChannels = htole16(num_channels);
|
||||
+ header.fmt.SampleRate = htole32(sample_rate);
|
||||
+ header.fmt.ByteRate =
|
||||
+ htole32(ByteRate(num_channels, sample_rate, bytes_per_sample));
|
||||
+ header.fmt.BlockAlign = htole16(BlockAlign(num_channels, bytes_per_sample));
|
||||
+ header.fmt.BitsPerSample = htole16(8 * bytes_per_sample);
|
||||
+ header.fmt.ExtensionSize = htole16(0);
|
||||
+ header.fact.header.ID = htole32(PackFourCC('f', 'a', 'c', 't'));
|
||||
+ header.fact.header.Size = htole32(4);
|
||||
+ header.fact.SampleLength = htole32(num_channels * num_samples);
|
||||
+ header.data.header.ID = htole32(PackFourCC('d', 'a', 't', 'a'));
|
||||
+ header.data.header.Size = htole32(bytes_in_payload);
|
||||
|
||||
// Do an extra copy rather than writing everything to buf directly, since buf
|
||||
// might not be correctly aligned.
|
||||
@@ -391,6 +403,7 @@ bool ReadWavHeader(WavHeaderReader* readable,
|
||||
return false;
|
||||
if (ReadFourCC(header.riff.Format) != "WAVE")
|
||||
return false;
|
||||
+ header.riff.header.Size = le32toh(header.riff.header.Size);
|
||||
|
||||
// Find "fmt " and "data" chunks. While the official Wave file specification
|
||||
// does not put requirements on the chunks order, it is uncommon to find the
|
||||
Loading…
Reference in a new issue