From ebc07ec32002c53702eb6e53ee1532ad2e0dc2bd Mon Sep 17 00:00:00 2001 From: Marcus Comstedt 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(-) diff --git a/third_party/webrtc/common_audio/wav_header.cc b/third_party/webrtc/common_audio/wav_header.cc index ce119f1095..3124f9e20c 100644 --- a/third_party/webrtc/common_audio/wav_header.cc +++ b/third_party/webrtc/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(a) << 24 | static_cast(b) << 16 | + static_cast(c) << 8 | static_cast(d); +#else uint32_t packed_value = static_cast(a) | static_cast(b) << 8 | static_cast(c) << 16 | static_cast(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(8 * bytes_per_sample); header.data.header.ID = PackFourCC('d', 'a', 't', 'a'); header.data.header.Size = static_cast(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(num_channels * num_samples); header.data.header.ID = PackFourCC('d', 'a', 't', 'a'); header.data.header.Size = static_cast(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 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(-) diff --git a/third_party/webrtc/common_audio/wav_file.cc b/third_party/webrtc/common_audio/wav_file.cc index e49126f139..80891bcb2d 100644 --- a/third_party/webrtc/common_audio/wav_file.cc +++ b/third_party/webrtc/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(static_cast(__builtin_bswap16(samples_to_convert[j]))); +#else samples[next_chunk_start + j] = static_cast(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 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 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 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 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