[patch] Webrtc patch from my chromium set

webrtc.patch (text/plain), 11.70 KB, created by Marcus Comstedt on 2022-10-01 07:24:58 UTC

From ebc07ec32002c53702eb6e53ee1532ad2e0dc2bd Mon Sep 17 00:00:00 2001
From: Marcus Comstedt <[e-mail]>
Date: Fri, 12 Mar 2021 23:27:16 +0100
Subject: [PATCH 1/2] wav: Swap header fields as needed

---
 third_party/libwebrtc/common_audio/wav_header.cc | 48 +++++++++++++++++--
 1 file changed, 44 insertions(+), 4 deletions(-)

diff --git a/third_party/libwebrtc/common_audio/wav_header.cc b/third_party/libwebrtc/common_audio/wav_header.cc
index ce119f1095..3124f9e20c 100644
--- a/third_party/libwebrtc/common_audio/wav_header.cc
+++ b/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 <[e-mail]>
Date: Fri, 12 Mar 2021 23:28:25 +0100
Subject: [PATCH 2/2] wav: Implement sample swapping

---
 third_party/libwebrtc/common_audio/wav_file.cc | 50 ++++++++++++++-------
 1 file changed, 34 insertions(+), 16 deletions(-)

diff --git a/third_party/libwebrtc/common_audio/wav_file.cc b/third_party/libwebrtc/common_audio/wav_file.cc
index e49126f139..80891bcb2d 100644
--- a/third_party/libwebrtc/common_audio/wav_file.cc
+++ b/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