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author | Piotr Kubaj <pkubaj@FreeBSD.org> | 2022-12-21 00:58:06 +0000 |
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committer | Piotr Kubaj <pkubaj@FreeBSD.org> | 2022-12-21 00:58:06 +0000 |
commit | 062cceb4bb8907b4be494343046efff348daf4cf (patch) | |
tree | d0455059d6a87373492806fd77800c8fce67464e | |
parent | 81eb9b8aaa98dd516ccdd0a141e3efa995c70d8e (diff) | |
download | ports-062cceb4bb8907b4be494343046efff348daf4cf.tar.gz ports-062cceb4bb8907b4be494343046efff348daf4cf.zip |
www/firefox-esr: enable webrtc on powerpc64
Patch copied from www/firefox/files/patch-libwebrtc-powerpc64.
-rw-r--r-- | www/firefox-esr/Makefile | 4 | ||||
-rw-r--r-- | www/firefox-esr/files/patch-libwebrtc-powerpc64 | 264 |
2 files changed, 266 insertions, 2 deletions
diff --git a/www/firefox-esr/Makefile b/www/firefox-esr/Makefile index 0217c562ec0d..534076d44353 100644 --- a/www/firefox-esr/Makefile +++ b/www/firefox-esr/Makefile @@ -1,6 +1,6 @@ PORTNAME= firefox DISTVERSION= 102.6.0 -PORTREVISION= 1 +PORTREVISION= 2 PORTEPOCH= 1 CATEGORIES= www wayland MASTER_SITES= MOZILLA/${PORTNAME}/releases/${DISTVERSION}esr/source \ @@ -48,7 +48,7 @@ MOZ_OPTIONS= --enable-application=browser \ .include <bsd.port.options.mk> .if ${ARCH} == powerpc64 -MOZ_OPTIONS+= --disable-webrtc --without-wasm-sandboxed-libraries +MOZ_OPTIONS+= --without-wasm-sandboxed-libraries .else BUILD_DEPENDS+= ${LOCALBASE}/share/wasi-sysroot/lib/wasm32-wasi/libc++abi.a:devel/wasi-libcxx \ ${LOCALBASE}/share/wasi-sysroot/lib/wasm32-wasi/libc.a:devel/wasi-libc \ diff --git a/www/firefox-esr/files/patch-libwebrtc-powerpc64 b/www/firefox-esr/files/patch-libwebrtc-powerpc64 new file mode 100644 index 000000000000..0dee667b1021 --- /dev/null +++ b/www/firefox-esr/files/patch-libwebrtc-powerpc64 @@ -0,0 +1,264 @@ +From ebc07ec32002c53702eb6e53ee1532ad2e0dc2bd Mon Sep 17 00:00:00 2001 +From: Marcus Comstedt <marcus@mc.pp.se> +Date: Fri, 12 Mar 2021 23:27:16 +0100 +Subject: [PATCH 1/2] wav: Swap header fields as needed + +--- + third_party/webrtc/common_audio/wav_header.cc | 48 +++++++++++++++++-- + 1 file changed, 44 insertions(+), 4 deletions(-) + +--- third_party/libwebrtc/common_audio/wav_header.cc ++++ third_party/libwebrtc/common_audio/wav_header.cc +@@ -26,10 +26,6 @@ + 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; +@@ -111,9 +107,15 @@ static_assert(sizeof(WavHeaderIeeeFloat) == kIeeeFloatWavHeaderSize, + "no padding in header"); + + uint32_t PackFourCC(char a, char b, char c, char d) { ++#ifndef WEBRTC_ARCH_LITTLE_ENDIAN ++ uint32_t packed_value = ++ static_cast<uint32_t>(a) << 24 | static_cast<uint32_t>(b) << 16 | ++ static_cast<uint32_t>(c) << 8 | static_cast<uint32_t>(d); ++#else + uint32_t packed_value = + static_cast<uint32_t>(a) | static_cast<uint32_t>(b) << 8 | + static_cast<uint32_t>(c) << 16 | static_cast<uint32_t>(d) << 24; ++#endif + return packed_value; + } + +@@ -172,6 +174,9 @@ bool FindWaveChunk(ChunkHeader* chunk_header, + if (readable->Read(chunk_header, sizeof(*chunk_header)) != + sizeof(*chunk_header)) + return false; // EOF. ++#ifndef WEBRTC_ARCH_LITTLE_ENDIAN ++ chunk_header->Size = __builtin_bswap32(chunk_header->Size); ++#endif + if (ReadFourCC(chunk_header->ID) == sought_chunk_id) + return true; // Sought chunk found. + // Ignore current chunk by skipping its payload. +@@ -185,6 +190,14 @@ bool ReadFmtChunkData(FmtPcmSubchunk* fmt_subchunk, WavHeaderReader* readable) { + if (readable->Read(&(fmt_subchunk->AudioFormat), kFmtPcmSubchunkSize) != + kFmtPcmSubchunkSize) + return false; ++#ifndef WEBRTC_ARCH_LITTLE_ENDIAN ++ fmt_subchunk->AudioFormat = __builtin_bswap16(fmt_subchunk->AudioFormat); ++ fmt_subchunk->NumChannels = __builtin_bswap16(fmt_subchunk->NumChannels); ++ fmt_subchunk->SampleRate = __builtin_bswap32(fmt_subchunk->SampleRate); ++ fmt_subchunk->ByteRate = __builtin_bswap32(fmt_subchunk->ByteRate); ++ fmt_subchunk->BlockAlign = __builtin_bswap16(fmt_subchunk->BlockAlign); ++ fmt_subchunk->BitsPerSample = __builtin_bswap16(fmt_subchunk->BitsPerSample); ++#endif + 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 +@@ -225,6 +238,17 @@ void WritePcmWavHeader(size_t num_channels, + header.fmt.BitsPerSample = static_cast<uint16_t>(8 * bytes_per_sample); + header.data.header.ID = PackFourCC('d', 'a', 't', 'a'); + header.data.header.Size = static_cast<uint32_t>(bytes_in_payload); ++#ifndef WEBRTC_ARCH_LITTLE_ENDIAN ++ header.riff.header.Size = __builtin_bswap32(header.riff.header.Size); ++ header.fmt.header.Size = __builtin_bswap32(header.fmt.header.Size); ++ header.fmt.AudioFormat = __builtin_bswap16(header.fmt.AudioFormat); ++ header.fmt.NumChannels = __builtin_bswap16(header.fmt.NumChannels); ++ header.fmt.SampleRate = __builtin_bswap32(header.fmt.SampleRate); ++ header.fmt.ByteRate = __builtin_bswap32(header.fmt.ByteRate); ++ header.fmt.BlockAlign = __builtin_bswap16(header.fmt.BlockAlign); ++ header.fmt.BitsPerSample = __builtin_bswap16(header.fmt.BitsPerSample); ++ header.data.header.Size = __builtin_bswap32(header.data.header.Size); ++#endif + + // Do an extra copy rather than writing everything to buf directly, since buf + // might not be correctly aligned. +@@ -261,6 +285,19 @@ void WriteIeeeFloatWavHeader(size_t num_channels, + 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); ++#ifndef WEBRTC_ARCH_LITTLE_ENDIAN ++ header.riff.header.Size = __builtin_bswap32(header.riff.header.Size); ++ header.fmt.header.Size = __builtin_bswap32(header.fmt.header.Size); ++ header.fmt.AudioFormat = __builtin_bswap16(header.fmt.AudioFormat); ++ header.fmt.NumChannels = __builtin_bswap16(header.fmt.NumChannels); ++ header.fmt.SampleRate = __builtin_bswap32(header.fmt.SampleRate); ++ header.fmt.ByteRate = __builtin_bswap32(header.fmt.ByteRate); ++ header.fmt.BlockAlign = __builtin_bswap16(header.fmt.BlockAlign); ++ header.fmt.BitsPerSample = __builtin_bswap16(header.fmt.BitsPerSample); ++ header.fact.header.Size = __builtin_bswap32(header.fact.header.Size); ++ header.fact.SampleLength = __builtin_bswap32(header.fact.SampleLength); ++ header.data.header.Size = __builtin_bswap32(header.data.header.Size); ++#endif + + // Do an extra copy rather than writing everything to buf directly, since buf + // might not be correctly aligned. +@@ -387,6 +424,9 @@ bool ReadWavHeader(WavHeaderReader* readable, + return false; + if (ReadFourCC(header.riff.Format) != "WAVE") + return false; ++#ifndef WEBRTC_ARCH_LITTLE_ENDIAN ++ header.riff.header.Size = __builtin_bswap32(header.riff.header.Size); ++#endif + + // 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 +-- +2.26.3 + + +From 28adaefe12a045a4adf7fdf56eb4e57db46dbe5e Mon Sep 17 00:00:00 2001 +From: Marcus Comstedt <marcus@mc.pp.se> +Date: Fri, 12 Mar 2021 23:28:25 +0100 +Subject: [PATCH 2/2] wav: Implement sample swapping + +--- + third_party/webrtc/common_audio/wav_file.cc | 50 ++++++++++++++------- + 1 file changed, 34 insertions(+), 16 deletions(-) + +--- third_party/libwebrtc/common_audio/wav_file.cc ++++ third_party/libwebrtc/common_audio/wav_file.cc +@@ -89,10 +89,6 @@ 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) { +@@ -107,6 +103,9 @@ size_t WavReader::ReadSamples(const size_t num_samples, + num_samples_read = num_bytes_read / sizeof(samples_to_convert[0]); + + for (size_t j = 0; j < num_samples_read; ++j) { ++#ifndef WEBRTC_ARCH_LITTLE_ENDIAN ++ *(uint32_t*)&samples_to_convert[j] = __builtin_bswap32(*(uint32_t*)&samples_to_convert[j]); ++#endif + samples[next_chunk_start + j] = FloatToS16(samples_to_convert[j]); + } + } else { +@@ -114,6 +113,11 @@ 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]); ++#ifndef WEBRTC_ARCH_LITTLE_ENDIAN ++ for (size_t j = 0; j < num_samples_read; ++j) { ++ samples[next_chunk_start + j] = __builtin_bswap16(samples[next_chunk_start + j]); ++ } ++#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 +133,6 @@ 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) { +@@ -147,8 +147,13 @@ size_t WavReader::ReadSamples(const size_t num_samples, float* const samples) { + num_samples_read = num_bytes_read / sizeof(samples_to_convert[0]); + + for (size_t j = 0; j < num_samples_read; ++j) { ++#ifndef WEBRTC_ARCH_LITTLE_ENDIAN ++ samples[next_chunk_start + j] = ++ static_cast<float>(static_cast<int16_t>(__builtin_bswap16(samples_to_convert[j]))); ++#else + samples[next_chunk_start + j] = + static_cast<float>(samples_to_convert[j]); ++#endif + } + } else { + RTC_CHECK_EQ(format_, WavFormat::kWavFormatIeeeFloat); +@@ -157,6 +162,9 @@ size_t WavReader::ReadSamples(const size_t num_samples, float* const samples) { + num_samples_read = num_bytes_read / sizeof(samples[0]); + + for (size_t j = 0; j < num_samples_read; ++j) { ++#ifndef WEBRTC_ARCH_LITTLE_ENDIAN ++ *(uint32_t*)&samples[next_chunk_start + j] = __builtin_bswap32(*(uint32_t*)&samples[next_chunk_start + j]); ++#endif + samples[next_chunk_start + j] = + FloatToFloatS16(samples[next_chunk_start + j]); + } +@@ -213,23 +221,31 @@ 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 ++ std::array<int16_t, kMaxChunksize> converted_samples; ++ for (size_t j = 0; j < num_samples_to_write; ++j) { ++ converted_samples[j] = __builtin_bswap16(samples[i + j]); ++ } ++ RTC_CHECK( ++ file_.Write(converted_samples.data(), num_samples_to_write * sizeof(samples[0]))); ++#else + RTC_CHECK( + file_.Write(&samples[i], num_samples_to_write * sizeof(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]); ++#ifndef WEBRTC_ARCH_LITTLE_ENDIAN ++ *(uint32_t*)&converted_samples[j] = __builtin_bswap32(*(uint32_t*)&converted_samples[j]); ++#endif + } + RTC_CHECK( + file_.Write(converted_samples.data(), +@@ -243,10 +259,6 @@ 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 = +@@ -256,6 +268,9 @@ void WavWriter::WriteSamples(const float* samples, size_t num_samples) { + 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]); ++#ifndef WEBRTC_ARCH_LITTLE_ENDIAN ++ converted_samples[j] = __builtin_bswap16(converted_samples[j]); ++#endif + } + RTC_CHECK( + file_.Write(converted_samples.data(), +@@ -265,6 +280,9 @@ void WavWriter::WriteSamples(const float* samples, size_t num_samples) { + std::array<float, kMaxChunksize> converted_samples; + for (size_t j = 0; j < num_samples_to_write; ++j) { + converted_samples[j] = FloatS16ToFloat(samples[i + j]); ++#ifndef WEBRTC_ARCH_LITTLE_ENDIAN ++ *(uint32_t*)&converted_samples[j] = __builtin_bswap32(*(uint32_t*)&converted_samples[j]); ++#endif + } + RTC_CHECK( + file_.Write(converted_samples.data(), +-- +2.26.3 + |