File big_endian_support.patch of Package webrtc-audio-processing-0

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File big_endian_support.patch of Package webrtc-audio-processing-0 (Revision 792aebf0b203020308221e0308c06db0)

Currently displaying revision 792aebf0b203020308221e0308c06db0 , Show latest

diff -up webrtc-audio-processing-0.2/webrtc/common_audio/wav_file.cc.than webrtc-audio-processing-0.2/webrtc/common_audio/wav_file.cc
--- webrtc-audio-processing-0.2/webrtc/common_audio/wav_file.cc.than	2016-05-24 08:28:45.749940095 -0400
+++ webrtc-audio-processing-0.2/webrtc/common_audio/wav_file.cc	2016-05-24 08:50:30.361020010 -0400
@@ -64,9 +64,6 @@ WavReader::~WavReader() {
 }
 
 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(rtc::checked_cast<uint32_t>(num_samples),
                          num_samples_remaining_);
@@ -76,6 +73,12 @@ size_t WavReader::ReadSamples(size_t num
   RTC_CHECK(read == num_samples || feof(file_handle_));
   RTC_CHECK_LE(read, num_samples_remaining_);
   num_samples_remaining_ -= rtc::checked_cast<uint32_t>(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;
 }
 
@@ -120,10 +123,17 @@ WavWriter::~WavWriter() {
 
 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
+  int16_t * le_samples = new int16_t[num_samples];
+  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_ += static_cast<uint32_t>(written);
   RTC_CHECK(written <= std::numeric_limits<uint32_t>::max() ||
diff -up webrtc-audio-processing-0.2/webrtc/common_audio/wav_header.cc.than webrtc-audio-processing-0.2/webrtc/common_audio/wav_header.cc
--- webrtc-audio-processing-0.2/webrtc/common_audio/wav_header.cc.than	2016-05-24 08:50:52.591379263 -0400
+++ webrtc-audio-processing-0.2/webrtc/common_audio/wav_header.cc	2016-05-24 08:52:08.552606848 -0400
@@ -129,7 +129,39 @@ static inline std::string ReadFourCC(uin
   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(uint32_t bytes_in_payload) {

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File big_endian_support.patch of Package webrtc-audio-processing-0 (Revision 792aebf0b203020308221e0308c06db0)

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