Merge changes I7bbf776e,I1b827825

am: fe8b4a6579

* commit 'fe8b4a657979b49e1701bd92f6d5814a99e0b2be': (7237 commits)
  WIP: Changes after merge commit 'cb3f9bd'
  Make the nonlinear beamformer steerable
  Utilize bitrate above codec max to protect video.
  Enable VP9 internal resize by default.
  Filter overlapping RTP header extensions.
  Make VCMEncodedFrameCallback const.
  MediaCodecVideoEncoder: Add number of quality resolution downscales to Encoded callback.
  Remove redudant encoder rate calls.
  Create isolate files for nonparallel tests.
  Register header extensions in RtpRtcpObserver to avoid log spam.
  Make an enum class out of NetEqDecoder, and hide the neteq_decoders_ table
  ACM: Move NACK functionality inside NetEq
  Fix chromium-style warnings in webrtc/sound/.
  Create a 'webrtc_nonparallel_tests' target.
  Update scalability structure data according to updates in the RTP payload profile.
  audio_coding: rename interface -> include
  Rewrote perform_action_on_all_files to be parallell.
  Update reference indices according to updates in the RTP payload profile.
  Disable P2PTransport...TestFailoverControlledSide on Memcheck
  pass clangcl compile options to ignore warnings in gflags.cc
  ...

1 files changed, 68 insertions, 0 deletions

diff --git a/webrtc/modules/audio_coding/main/test/PCMFile.h b/webrtc/modules/audio_coding/main/test/PCMFile.h
new file mode 100644
index 0000000000..8353898f03
--- /dev/null
+++ b/webrtc/modules/audio_coding/main/test/PCMFile.h
@@ -0,0 +1,68 @@
+/*
+ *  Copyright (c) 2012 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.
+ */
+
+#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_PCMFILE_H_
+#define WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_PCMFILE_H_
+
+#include <stdio.h>
+#include <stdlib.h>
+
+#include <string>
+
+#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/typedefs.h"
+
+namespace webrtc {
+
+class PCMFile {
+ public:
+  PCMFile();
+  PCMFile(uint32_t timestamp);
+  ~PCMFile() {
+    if (pcm_file_ != NULL) {
+      fclose(pcm_file_);
+    }
+  }
+
+  void Open(const std::string& filename, uint16_t frequency, const char* mode,
+            bool auto_rewind = false);
+
+  int32_t Read10MsData(AudioFrame& audio_frame);
+
+  void Write10MsData(int16_t *playout_buffer, size_t length_smpls);
+  void Write10MsData(AudioFrame& audio_frame);
+
+  uint16_t PayloadLength10Ms() const;
+  int32_t SamplingFrequency() const;
+  void Close();
+  bool EndOfFile() const {
+    return end_of_file_;
+  }
+  void Rewind();
+  static int16_t ChooseFile(std::string* file_name, int16_t max_len,
+                            uint16_t* frequency_hz);
+  bool Rewinded();
+  void SaveStereo(bool is_stereo = true);
+  void ReadStereo(bool is_stereo = true);
+ private:
+  FILE* pcm_file_;
+  uint16_t samples_10ms_;
+  int32_t frequency_;
+  bool end_of_file_;
+  bool auto_rewind_;
+  bool rewinded_;
+  uint32_t timestamp_;
+  bool read_stereo_;
+  bool save_stereo_;
+};
+
+}  // namespace webrtc
+
+#endif  // WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_PCMFILE_H_