1 files changed, 90 insertions, 0 deletions
diff --git a/chromium/third_party/webrtc/modules/audio_coding/codecs/tools/audio_codec_speed_test.h b/chromium/third_party/webrtc/modules/audio_coding/codecs/tools/audio_codec_speed_test.h
new file mode 100644
index 00000000000..2c9b45e4f86
--- /dev/null
+++ b/chromium/third_party/webrtc/modules/audio_coding/codecs/tools/audio_codec_speed_test.h
@@ -0,0 +1,90 @@
+/*
+ *  Copyright (c) 2014 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_CODECS_TOOLS_AUDIO_CODEC_SPEED_TEST_H_
+#define WEBRTC_MODULES_AUDIO_CODING_CODECS_TOOLS_AUDIO_CODEC_SPEED_TEST_H_
+
+#include <string>
+#include "testing/gtest/include/gtest/gtest.h"
+#include "webrtc/system_wrappers/interface/scoped_ptr.h"
+#include "webrtc/typedefs.h"
+
+namespace webrtc {
+
+// Define coding parameter as
+// <channels, bit_rate, file_name, extension, if_save_output>.
+typedef std::tr1::tuple<int, int, std::string, std::string, bool> coding_param;
+
+class AudioCodecSpeedTest : public testing::TestWithParam<coding_param> {
+ protected:
+  AudioCodecSpeedTest(int block_duration_ms,
+                      int input_sampling_khz,
+                      int output_sampling_khz);
+  virtual void SetUp();
+  virtual void TearDown();
+
+  // EncodeABlock(...) does the following:
+  // 1. encodes a block of audio, saved in |in_data|,
+  // 2. save the bit stream to |bit_stream| of |max_bytes| bytes in size,
+  // 3. assign |encoded_bytes| with the length of the bit stream (in bytes),
+  // 4. return the cost of time (in millisecond) spent on actual encoding.
+  virtual float EncodeABlock(int16_t* in_data, uint8_t* bit_stream,
+                             int max_bytes, int* encoded_bytes) = 0;
+
+  // DecodeABlock(...) does the following:
+  // 1. decodes the bit stream in |bit_stream| with a length of |encoded_bytes|
+  // (in bytes),
+  // 2. save the decoded audio in |out_data|,
+  // 3. return the cost of time (in millisecond) spent on actual decoding.
+  virtual float DecodeABlock(const uint8_t* bit_stream, int encoded_bytes,
+                             int16_t* out_data) = 0;
+
+  // Encoding and decode an audio of |audio_duration| (in seconds) and
+  // record the runtime for encoding and decoding separately.
+  void EncodeDecode(size_t audio_duration);
+
+  int block_duration_ms_;
+  int input_sampling_khz_;
+  int output_sampling_khz_;
+
+  // Number of samples-per-channel in a frame.
+  int input_length_sample_;
+
+  // Expected output number of samples-per-channel in a frame.
+  int output_length_sample_;
+
+  scoped_ptr<int16_t[]> in_data_;
+  scoped_ptr<int16_t[]> out_data_;
+  size_t data_pointer_;
+  size_t loop_length_samples_;
+  scoped_ptr<uint8_t[]> bit_stream_;
+
+  // Maximum number of bytes in output bitstream for a frame of audio.
+  int max_bytes_;
+
+  int encoded_bytes_;
+  float encoding_time_ms_;
+  float decoding_time_ms_;
+  FILE* out_file_;
+
+  int channels_;
+
+  // Bit rate is in bit-per-second.
+  int bit_rate_;
+
+  std::string in_filename_;
+
+  // Determines whether to save the output to file.
+  bool save_out_data_;
+};
+
+}  // namespace webrtc
+
+#endif  // WEBRTC_MODULES_AUDIO_CODING_CODECS_TOOLS_AUDIO_CODEC_SPEED_TEST_H_