Fixed build of libtgvoip on Big-Endian architectures.

Telegram/ThirdParty/libtgvoip/webrtc_dsp/webrtc/common_audio/wav_file.cc

@@ -77,9 +77,6 @@ size_t WavReader::num_samples() const {
 }
 
 size_t WavReader::ReadSamples(size_t num_samples, int16_t* samples) {
-#ifndef WEBRTC_ARCH_LITTLE_ENDIAN
-#error "Need to convert samples to big-endian when reading from WAV file"
-#endif
   // There could be metadata after the audio; ensure we don't read it.
   num_samples = std::min(num_samples, num_samples_remaining_);
   const size_t read =
@@ -88,6 +85,12 @@ size_t WavReader::ReadSamples(size_t num_samples, int16_t* samples) {
   RTC_CHECK(read == num_samples || feof(file_handle_));
   RTC_CHECK_LE(read, num_samples_remaining_);
   num_samples_remaining_ -= read;
+#ifndef WEBRTC_ARCH_LITTLE_ENDIAN
+  //convert to big-endian
+  for(size_t idx = 0; idx < num_samples; idx++) {
+    samples[idx] = (samples[idx]<<8) | (samples[idx]>>8);
+  }
+#endif
   return read;
 }
 
@@ -144,10 +147,17 @@ size_t WavWriter::num_samples() const {
 
 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"
+  int16_t * le_samples = new int16_t[num_samples];
-#endif
+  for(size_t idx = 0; idx < num_samples; idx++) {
+    le_samples[idx] = (samples[idx]<<8) | (samples[idx]>>8);
+  }
+  const size_t written =
+      fwrite(le_samples, sizeof(*le_samples), num_samples, file_handle_);
+  delete []le_samples;
+#else
   const size_t written =
       fwrite(samples, sizeof(*samples), num_samples, file_handle_);
+#endif
   RTC_CHECK_EQ(num_samples, written);
   num_samples_ += written;
   RTC_CHECK(num_samples_ >= written);  // detect size_t overflow

Telegram/ThirdParty/libtgvoip/webrtc_dsp/webrtc/common_audio/wav_header.cc

@@ -127,7 +127,39 @@ static inline std::string ReadFourCC(uint32_t x) {
   return std::string(reinterpret_cast<char*>(&x), 4);
 }
 #else
-#error "Write be-to-le conversion functions"
+static inline void WriteLE16(uint16_t* f, uint16_t x) {
+  *f = ((x << 8) & 0xff00)  | ( ( x >> 8) & 0x00ff);
+}
+
+static inline void WriteLE32(uint32_t* f, uint32_t x) {
+    *f = ( (x & 0x000000ff) << 24 )
+      | ((x & 0x0000ff00) << 8)
+      | ((x & 0x00ff0000) >> 8)
+      | ((x & 0xff000000) >> 24 );
+}
+
+static inline void WriteFourCC(uint32_t* f, char a, char b, char c, char d) {
+    *f = (static_cast<uint32_t>(a) << 24 )
+      |  (static_cast<uint32_t>(b) << 16)
+      |  (static_cast<uint32_t>(c) << 8)
+      |  (static_cast<uint32_t>(d) );
+}
+
+static inline uint16_t ReadLE16(uint16_t x) {
+  return  (( x & 0x00ff) << 8 )| ((x & 0xff00)>>8);
+}
+
+static inline uint32_t ReadLE32(uint32_t x) {
+  return   ( (x & 0x000000ff) << 24 )
+         | ( (x & 0x0000ff00) << 8 )
+         | ( (x & 0x00ff0000) >> 8)
+         | ( (x & 0xff000000) >> 24 );
+}
+
+static inline std::string ReadFourCC(uint32_t x) {
+  x = ReadLE32(x);
+  return std::string(reinterpret_cast<char*>(&x), 4);
+}
 #endif
 
 static inline uint32_t RiffChunkSize(size_t bytes_in_payload) {

Telegram/ThirdParty/libtgvoip/webrtc_dsp/webrtc/typedefs.h

@@ -48,7 +48,19 @@
 #define WEBRTC_ARCH_32_BITS
 #define WEBRTC_ARCH_LITTLE_ENDIAN
 #else
-#error Please add support for your architecture in typedefs.h
+/* instead of failing, use typical unix defines... */
+#if __BYTE_ORDER__ == __ORDER_LITTLE_ENDIAN__
+#define WEBRTC_ARCH_LITTLE_ENDIAN
+#elif __BYTE_ORDER__ == __ORDER_BIG_ENDIAN__
+#define WEBRTC_ARCH_BIG_ENDIAN
+#else
+#error __BYTE_ORDER__ is not defined
+#endif
+#if defined(__LP64__)
+#define WEBRTC_ARCH_64_BITS
+#else
+#define WEBRTC_ARCH_32_BITS
+#endif
 #endif
 
 #if !(defined(WEBRTC_ARCH_LITTLE_ENDIAN) ^ defined(WEBRTC_ARCH_BIG_ENDIAN))