diff options
Diffstat (limited to 'media-libs/webrtc-audio-processing/files/webrtc-audio-processing-1.3-big-endian-support.patch')
| -rw-r--r-- | media-libs/webrtc-audio-processing/files/webrtc-audio-processing-1.3-big-endian-support.patch | 324 |
1 files changed, 324 insertions, 0 deletions
diff --git a/media-libs/webrtc-audio-processing/files/webrtc-audio-processing-1.3-big-endian-support.patch b/media-libs/webrtc-audio-processing/files/webrtc-audio-processing-1.3-big-endian-support.patch new file mode 100644 index 000000000000..3984cf70124c --- /dev/null +++ b/media-libs/webrtc-audio-processing/files/webrtc-audio-processing-1.3-big-endian-support.patch @@ -0,0 +1,324 @@ +https://bugs.gentoo.org/917493 +https://sources.debian.org/src/webrtc-audio-processing/1.0-0.2/debian/patches/big-endian-support.patch/ + +Description: big endian support + Provide endianness converters before writing or after reading WAV +Author: Nicholas Guriev <nicholas@guriev.su> +Bug-telegram: https://github.com/desktop-app/tg_owt/pull/46 +Origin: https://github.com/desktop-app/tg_owt/commit/65f002e +Last-Update: 2022-02-01 +--- +This patch header follows DEP-3: http://dep.debian.net/deps/dep3/ +--- 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 <byteswap.h> + #include <errno.h> + + #include <algorithm> +@@ -34,6 +35,38 @@ + format == WavFormat::kWavFormatIeeeFloat; + } + ++template <typename T> ++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<const uint16_t*>(source); ++ uint16_t* dst = reinterpret_cast<uint16_t*>(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<const uint32_t*>(source); ++ uint32_t* dst = reinterpret_cast<uint32_t*>(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<const uint64_t*>(source); ++ uint64_t* dst = reinterpret_cast<uint64_t*>(destination); ++ for (size_t index = 0; index < length; index++) { ++ dst[index] = bswap_64(src[index]); ++ } ++ } ++} ++ ++template <typename T> ++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 @@ + + 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 @@ + 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 @@ + 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, 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 @@ + 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 @@ + 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 @@ + } + + 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<int16_t, kMaxChunksize> 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<float, kMaxChunksize> 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 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 @@ + 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 @@ + 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]))); +--- a/webrtc/common_audio/wav_header.cc ++++ b/webrtc/common_audio/wav_header.cc +@@ -14,6 +14,8 @@ + + #include "common_audio/wav_header.h" + ++#include <endian.h> ++ + #include <cstring> + #include <limits> + #include <string> +@@ -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; +@@ -172,6 +170,8 @@ + 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. +@@ -185,6 +185,13 @@ + 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 +@@ -212,19 +219,22 @@ + auto header = rtc::MsanUninitialized<WavHeaderPcm>({}); + 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<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.data.header.ID = PackFourCC('d', 'a', 't', 'a'); +- header.data.header.Size = static_cast<uint32_t>(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. +@@ -243,24 +253,26 @@ + auto header = rtc::MsanUninitialized<WavHeaderIeeeFloat>({}); + 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<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. +@@ -389,6 +401,7 @@ + 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 |