From c1369d1b277be895d7c7f9e1defa8465e3349abe Mon Sep 17 00:00:00 2001 From: Maxim Cournoyer Date: Wed, 24 Jan 2024 12:07:58 -0500 Subject: 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 --- gnu/local.mk | 1 - gnu/packages/audio.scm | 29 +- .../webrtc-audio-processing-big-endian.patch | 331 --------------------- 3 files changed, 2 insertions(+), 359 deletions(-) delete mode 100644 gnu/packages/patches/webrtc-audio-processing-big-endian.patch diff --git a/gnu/local.mk b/gnu/local.mk index 4f41e14867..f193163f14 100644 --- a/gnu/local.mk +++ b/gnu/local.mk @@ -2201,7 +2201,6 @@ dist_patch_DATA = \ %D%/packages/patches/wcstools-extend-makefiles.patch \ %D%/packages/patches/wdl-link-libs-and-fix-jnetlib.patch \ %D%/packages/patches/webkitgtk-adjust-bubblewrap-paths.patch \ - %D%/packages/patches/webrtc-audio-processing-big-endian.patch \ %D%/packages/patches/webrtc-for-telegram-desktop-unbundle-libsrtp.patch \ %D%/packages/patches/websocketpp-fix-for-cmake-3.15.patch \ %D%/packages/patches/wmctrl-64-fix.patch \ diff --git a/gnu/packages/audio.scm b/gnu/packages/audio.scm index 88e4dd2f3c..1c8ceb11a9 100644 --- a/gnu/packages/audio.scm +++ b/gnu/packages/audio.scm @@ -273,34 +273,9 @@ (define-public webrtc-audio-processing (string-append "http://freedesktop.org/software/pulseaudio/" name "/" name "-" version ".tar.gz")) (sha256 - (base32 "0xfvq5lxg612vfzk3zk6896zcb4cgrrb7fq76w9h40magz0jymcm")) - (modules '((guix build utils))) - (snippet - #~(begin - ;; See: - ;; . - (substitute* "meson.build" - (("absl_flags_registry") "absl_flags_reflection")) - (substitute* "webrtc/rtc_base/system/arch.h" - (("defined\\(__aarch64__\\)" all) - (string-append - ;; powerpc-linux - "(defined(__PPC__) && __SIZEOF_SIZE_T__ == 4)\n" - "#define WEBRTC_ARCH_32_BITS\n" - "#define WEBRTC_ARCH_BIG_ENDIAN\n" - ;; powerpc64-linux - "#elif (defined(__PPC64__) && defined(_BIG_ENDIAN))\n" - "#define WEBRTC_ARCH_64_BITS\n" - "#define WEBRTC_ARCH_BIG_ENDIAN\n" - ;; aarch64-linux - "#elif " all - ;; riscv64-linux - " || (defined(__riscv) && __riscv_xlen == 64)" - ;; powerpc64le-linux - " || (defined(__PPC64__) && defined(_LITTLE_ENDIAN))"))))) - (patches - (search-patches "webrtc-audio-processing-big-endian.patch")))) + (base32 "0xfvq5lxg612vfzk3zk6896zcb4cgrrb7fq76w9h40magz0jymcm")))) (build-system meson-build-system) + (native-inputs (list pkg-config)) (inputs (list abseil-cpp)) (synopsis "WebRTC's Audio Processing Library") (description "WebRTC-Audio-Processing library based on Google's diff --git a/gnu/packages/patches/webrtc-audio-processing-big-endian.patch b/gnu/packages/patches/webrtc-audio-processing-big-endian.patch deleted file mode 100644 index 1690597025..0000000000 --- a/gnu/packages/patches/webrtc-audio-processing-big-endian.patch +++ /dev/null @@ -1,331 +0,0 @@ -https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/issues/127 -https://github.com/desktop-app/tg_owt/commit/65f002e - -From 65f002eeda1d97ddc70c8c49ec563987203c76f5 Mon Sep 17 00:00:00 2001 -From: Nicholas Guriev -Date: Thu, 28 Jan 2021 20:54:06 +0300 -Subject: [PATCH] Provide endianness converters before writing or after reading - WAV - ---- - src/common_audio/wav_file.cc | 80 ++++++++++++++++++++++++++------- - src/common_audio/wav_header.cc | 81 ++++++++++++++++++++-------------- - 2 files changed, 111 insertions(+), 50 deletions(-) - -diff --git a/src/common_audio/wav_file.cc b/src/common_audio/wav_file.cc -index e49126f1..b5292668 100644 ---- a/webrtc/common_audio/wav_file.cc -+++ b/webrtc/common_audio/wav_file.cc -@@ -10,6 +10,7 @@ - - #include "common_audio/wav_file.h" - -+#include - #include - - #include -@@ -34,6 +35,38 @@ bool FormatSupported(WavFormat format) { - format == WavFormat::kWavFormatIeeeFloat; - } - -+template -+void TranslateEndianness(T* destination, const T* source, size_t length) { -+ static_assert(sizeof(T) == 2 || sizeof(T) == 4 || sizeof(T) == 8, -+ "no converter, use integral types"); -+ if (sizeof(T) == 2) { -+ const uint16_t* src = reinterpret_cast(source); -+ uint16_t* dst = reinterpret_cast(destination); -+ for (size_t index = 0; index < length; index++) { -+ dst[index] = bswap_16(src[index]); -+ } -+ } -+ if (sizeof(T) == 4) { -+ const uint32_t* src = reinterpret_cast(source); -+ uint32_t* dst = reinterpret_cast(destination); -+ for (size_t index = 0; index < length; index++) { -+ dst[index] = bswap_32(src[index]); -+ } -+ } -+ if (sizeof(T) == 8) { -+ const uint64_t* src = reinterpret_cast(source); -+ uint64_t* dst = reinterpret_cast(destination); -+ for (size_t index = 0; index < length; index++) { -+ dst[index] = bswap_64(src[index]); -+ } -+ } -+} -+ -+template -+void TranslateEndianness(T* buffer, size_t length) { -+ TranslateEndianness(buffer, buffer, length); -+} -+ - // Doesn't take ownership of the file handle and won't close it. - class WavHeaderFileReader : public WavHeaderReader { - public: -@@ -89,10 +122,6 @@ 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) { -@@ -105,6 +134,9 @@ size_t WavReader::ReadSamples(const size_t num_samples, - num_bytes_read = file_.Read(samples_to_convert.data(), - chunk_size * sizeof(samples_to_convert[0])); - num_samples_read = num_bytes_read / sizeof(samples_to_convert[0]); -+#ifdef WEBRTC_ARCH_BIG_ENDIAN -+ TranslateEndianness(samples_to_convert.data(), num_samples_read); -+#endif - - for (size_t j = 0; j < num_samples_read; ++j) { - samples[next_chunk_start + j] = FloatToS16(samples_to_convert[j]); -@@ -114,6 +146,10 @@ size_t WavReader::ReadSamples(const size_t num_samples, - num_bytes_read = file_.Read(&samples[next_chunk_start], - chunk_size * sizeof(samples[0])); - num_samples_read = num_bytes_read / sizeof(samples[0]); -+ -+#ifdef WEBRTC_ARCH_BIG_ENDIAN -+ TranslateEndianness(&samples[next_chunk_start], num_samples_read); -+#endif - } - RTC_CHECK(num_samples_read == 0 || (num_bytes_read % num_samples_read) == 0) - << "Corrupt file: file ended in the middle of a sample."; -@@ -129,10 +165,6 @@ size_t WavReader::ReadSamples(const size_t num_samples, - } - - 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) { -@@ -145,6 +177,9 @@ size_t WavReader::ReadSamples(const size_t num_samples, float* const samples) { - num_bytes_read = file_.Read(samples_to_convert.data(), - chunk_size * sizeof(samples_to_convert[0])); - num_samples_read = num_bytes_read / sizeof(samples_to_convert[0]); -+#ifdef WEBRTC_ARCH_BIG_ENDIAN -+ TranslateEndianness(samples_to_convert.data(), num_samples_read); -+#endif - - for (size_t j = 0; j < num_samples_read; ++j) { - samples[next_chunk_start + j] = -@@ -155,6 +190,9 @@ size_t WavReader::ReadSamples(const size_t num_samples, float* const samples) { - num_bytes_read = file_.Read(&samples[next_chunk_start], - chunk_size * sizeof(samples[0])); - num_samples_read = num_bytes_read / sizeof(samples[0]); -+#ifdef WEBRTC_ARCH_BIG_ENDIAN -+ TranslateEndianness(&samples[next_chunk_start], num_samples_read); -+#endif - - for (size_t j = 0; j < num_samples_read; ++j) { - samples[next_chunk_start + j] = -@@ -213,24 +251,32 @@ 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_BIG_ENDIAN - RTC_CHECK( - file_.Write(&samples[i], num_samples_to_write * sizeof(samples[0]))); -+#else -+ std::array converted_samples; -+ TranslateEndianness(converted_samples.data(), &samples[i], -+ num_samples_to_write); -+ 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 converted_samples; - for (size_t j = 0; j < num_samples_to_write; ++j) { - converted_samples[j] = S16ToFloat(samples[i + j]); - } -+#ifdef WEBRTC_ARCH_BIG_ENDIAN -+ TranslateEndianness(converted_samples.data(), num_samples_to_write); -+#endif - RTC_CHECK( - file_.Write(converted_samples.data(), - num_samples_to_write * sizeof(converted_samples[0]))); -@@ -243,10 +289,6 @@ 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 = -@@ -257,6 +299,9 @@ void WavWriter::WriteSamples(const float* samples, size_t num_samples) { - for (size_t j = 0; j < num_samples_to_write; ++j) { - converted_samples[j] = FloatS16ToS16(samples[i + j]); - } -+#ifdef WEBRTC_ARCH_BIG_ENDIAN -+ TranslateEndianness(converted_samples.data(), num_samples_to_write); -+#endif - RTC_CHECK( - file_.Write(converted_samples.data(), - num_samples_to_write * sizeof(converted_samples[0]))); -@@ -266,6 +311,9 @@ void WavWriter::WriteSamples(const float* samples, size_t num_samples) { - for (size_t j = 0; j < num_samples_to_write; ++j) { - converted_samples[j] = FloatS16ToFloat(samples[i + j]); - } -+#ifdef WEBRTC_ARCH_BIG_ENDIAN -+ TranslateEndianness(converted_samples.data(), num_samples_to_write); -+#endif - RTC_CHECK( - file_.Write(converted_samples.data(), - num_samples_to_write * sizeof(converted_samples[0]))); -diff --git a/webrtc/common_audio/wav_header.cc b/webrtc/common_audio/wav_header.cc -index 1ccbffca..98264a5c 100644 ---- a/src/common_audio/wav_header.cc -+++ b/src/common_audio/wav_header.cc -@@ -14,6 +14,8 @@ - - #include "common_audio/wav_header.h" - -+#include -+ - #include - #include - #include -@@ -26,10 +28,6 @@ - namespace webrtc { - namespace { - --#ifndef WEBRTC_ARCH_LITTLE_ENDIAN --#error "Code not working properly for big endian platforms." --#endif -- - #pragma pack(2) - struct ChunkHeader { - uint32_t ID; -@@ -174,6 +172,8 @@ bool FindWaveChunk(ChunkHeader* chunk_header, - if (readable->Read(chunk_header, sizeof(*chunk_header)) != - sizeof(*chunk_header)) - return false; // EOF. -+ chunk_header->Size = le32toh(chunk_header->Size); -+ - if (ReadFourCC(chunk_header->ID) == sought_chunk_id) - return true; // Sought chunk found. - // Ignore current chunk by skipping its payload. -@@ -187,6 +187,13 @@ bool ReadFmtChunkData(FmtPcmSubchunk* fmt_subchunk, WavHeaderReader* readable) { - if (readable->Read(&(fmt_subchunk->AudioFormat), kFmtPcmSubchunkSize) != - kFmtPcmSubchunkSize) - return false; -+ fmt_subchunk->AudioFormat = le16toh(fmt_subchunk->AudioFormat); -+ fmt_subchunk->NumChannels = le16toh(fmt_subchunk->NumChannels); -+ fmt_subchunk->SampleRate = le32toh(fmt_subchunk->SampleRate); -+ fmt_subchunk->ByteRate = le32toh(fmt_subchunk->ByteRate); -+ fmt_subchunk->BlockAlign = le16toh(fmt_subchunk->BlockAlign); -+ fmt_subchunk->BitsPerSample = le16toh(fmt_subchunk->BitsPerSample); -+ - const uint32_t fmt_size = fmt_subchunk->header.Size; - if (fmt_size != kFmtPcmSubchunkSize) { - // There is an optional two-byte extension field permitted to be present -@@ -214,19 +221,22 @@ void WritePcmWavHeader(size_t num_channels, - auto header = rtc::MsanUninitialized({}); - const size_t bytes_in_payload = bytes_per_sample * num_samples; - -- header.riff.header.ID = PackFourCC('R', 'I', 'F', 'F'); -- header.riff.header.Size = RiffChunkSize(bytes_in_payload, *header_size); -- header.riff.Format = PackFourCC('W', 'A', 'V', 'E'); -- header.fmt.header.ID = PackFourCC('f', 'm', 't', ' '); -- header.fmt.header.Size = kFmtPcmSubchunkSize; -- header.fmt.AudioFormat = MapWavFormatToHeaderField(WavFormat::kWavFormatPcm); -- header.fmt.NumChannels = static_cast(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(8 * bytes_per_sample); -- header.data.header.ID = PackFourCC('d', 'a', 't', 'a'); -- header.data.header.Size = static_cast(bytes_in_payload); -+ header.riff.header.ID = htole32(PackFourCC('R', 'I', 'F', 'F')); -+ header.riff.header.Size = -+ htole32(RiffChunkSize(bytes_in_payload, *header_size)); -+ header.riff.Format = htole32(PackFourCC('W', 'A', 'V', 'E')); -+ header.fmt.header.ID = htole32(PackFourCC('f', 'm', 't', ' ')); -+ header.fmt.header.Size = htole32(kFmtPcmSubchunkSize); -+ header.fmt.AudioFormat = -+ htole16(MapWavFormatToHeaderField(WavFormat::kWavFormatPcm)); -+ 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.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. -@@ -245,24 +255,26 @@ void WriteIeeeFloatWavHeader(size_t num_channels, - auto header = rtc::MsanUninitialized({}); - const size_t bytes_in_payload = bytes_per_sample * num_samples; - -- header.riff.header.ID = PackFourCC('R', 'I', 'F', 'F'); -- header.riff.header.Size = RiffChunkSize(bytes_in_payload, *header_size); -- header.riff.Format = PackFourCC('W', 'A', 'V', 'E'); -- header.fmt.header.ID = PackFourCC('f', 'm', 't', ' '); -- header.fmt.header.Size = kFmtIeeeFloatSubchunkSize; -+ header.riff.header.ID = htole32(PackFourCC('R', 'I', 'F', 'F')); -+ header.riff.header.Size = -+ htole32(RiffChunkSize(bytes_in_payload, *header_size)); -+ header.riff.Format = htole32(PackFourCC('W', 'A', 'V', 'E')); -+ header.fmt.header.ID = htole32(PackFourCC('f', 'm', 't', ' ')); -+ header.fmt.header.Size = htole32(kFmtIeeeFloatSubchunkSize); - header.fmt.AudioFormat = -- MapWavFormatToHeaderField(WavFormat::kWavFormatIeeeFloat); -- header.fmt.NumChannels = static_cast(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(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(num_channels * num_samples); -- header.data.header.ID = PackFourCC('d', 'a', 't', 'a'); -- header.data.header.Size = static_cast(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 -- cgit v1.2.3