1 files changed, 96 insertions, 0 deletions
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/tools/input_audio_file.cc b/third_party/libwebrtc/modules/audio_coding/neteq/tools/input_audio_file.cc
new file mode 100644
index 0000000000..b077dbff21
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/tools/input_audio_file.cc
@@ -0,0 +1,96 @@
+/*
+ *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_coding/neteq/tools/input_audio_file.h"
+
+#include "absl/strings/string_view.h"
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+namespace test {
+
+InputAudioFile::InputAudioFile(absl::string_view file_name, bool loop_at_end)
+    : loop_at_end_(loop_at_end) {
+  fp_ = fopen(std::string(file_name).c_str(), "rb");
+  RTC_DCHECK(fp_) << file_name << " could not be opened.";
+}
+
+InputAudioFile::~InputAudioFile() {
+  RTC_DCHECK(fp_);
+  fclose(fp_);
+}
+
+bool InputAudioFile::Read(size_t samples, int16_t* destination) {
+  if (!fp_) {
+    return false;
+  }
+  size_t samples_read = fread(destination, sizeof(int16_t), samples, fp_);
+  if (samples_read < samples) {
+    if (!loop_at_end_) {
+      return false;
+    }
+    // Rewind and read the missing samples.
+    rewind(fp_);
+    size_t missing_samples = samples - samples_read;
+    if (fread(destination + samples_read, sizeof(int16_t), missing_samples,
+              fp_) < missing_samples) {
+      // Could not read enough even after rewinding the file.
+      return false;
+    }
+  }
+  return true;
+}
+
+bool InputAudioFile::Seek(int samples) {
+  if (!fp_) {
+    return false;
+  }
+  // Find file boundaries.
+  const long current_pos = ftell(fp_);
+  RTC_CHECK_NE(EOF, current_pos)
+      << "Error returned when getting file position.";
+  RTC_CHECK_EQ(0, fseek(fp_, 0, SEEK_END));  // Move to end of file.
+  const long file_size = ftell(fp_);
+  RTC_CHECK_NE(EOF, file_size) << "Error returned when getting file position.";
+  // Find new position.
+  long new_pos = current_pos + sizeof(int16_t) * samples;  // Samples to bytes.
+  if (loop_at_end_) {
+    new_pos = new_pos % file_size;  // Wrap around the end of the file.
+    if (new_pos < 0) {
+      // For negative values of new_pos, newpos % file_size will also be
+      // negative. To get the correct result it's needed to add file_size.
+      new_pos += file_size;
+    }
+  } else {
+    new_pos = new_pos > file_size ? file_size : new_pos;  // Don't loop.
+  }
+  RTC_CHECK_GE(new_pos, 0)
+      << "Trying to move to before the beginning of the file";
+  // Move to new position relative to the beginning of the file.
+  RTC_CHECK_EQ(0, fseek(fp_, new_pos, SEEK_SET));
+  return true;
+}
+
+void InputAudioFile::DuplicateInterleaved(const int16_t* source,
+                                          size_t samples,
+                                          size_t channels,
+                                          int16_t* destination) {
+  // Start from the end of `source` and `destination`, and work towards the
+  // beginning. This is to allow in-place interleaving of the same array (i.e.,
+  // `source` and `destination` are the same array).
+  for (int i = static_cast<int>(samples - 1); i >= 0; --i) {
+    for (int j = static_cast<int>(channels - 1); j >= 0; --j) {
+      destination[i * channels + j] = source[i];
+    }
+  }
+}
+
+}  // namespace test
+}  // namespace webrtc