Moving src/webrtc into src/.

In order to eliminate the WebRTC Subtree mirror in Chromium, 
WebRTC is moving the content of the src/webrtc directory up
to the src/ directory.

NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
TBR=tommi@webrtc.org

Bug: chromium:611808
Change-Id: Iac59c5b51b950f174119565bac87955a7994bc38
Reviewed-on: https://webrtc-review.googlesource.com/1560
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Henrik Kjellander <kjellander@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19845}

modules/audio_coding/codecs/tools/audio_codec_speed_test.cc

@@ -0,0 +1,127 @@
+/*
+ *  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.
+ */
+
+#include "webrtc/modules/audio_coding/codecs/tools/audio_codec_speed_test.h"
+
+#include "webrtc/rtc_base/format_macros.h"
+#include "webrtc/test/gtest.h"
+#include "webrtc/test/testsupport/fileutils.h"
+
+using ::std::tr1::get;
+
+namespace webrtc {
+
+AudioCodecSpeedTest::AudioCodecSpeedTest(int block_duration_ms,
+                                         int input_sampling_khz,
+                                         int output_sampling_khz)
+    : block_duration_ms_(block_duration_ms),
+      input_sampling_khz_(input_sampling_khz),
+      output_sampling_khz_(output_sampling_khz),
+      input_length_sample_(
+          static_cast<size_t>(block_duration_ms_ * input_sampling_khz_)),
+      output_length_sample_(
+          static_cast<size_t>(block_duration_ms_ * output_sampling_khz_)),
+      data_pointer_(0),
+      loop_length_samples_(0),
+      max_bytes_(0),
+      encoded_bytes_(0),
+      encoding_time_ms_(0.0),
+      decoding_time_ms_(0.0),
+      out_file_(NULL) {
+}
+
+void AudioCodecSpeedTest::SetUp() {
+  channels_ = get<0>(GetParam());
+  bit_rate_ = get<1>(GetParam());
+  in_filename_ = test::ResourcePath(get<2>(GetParam()), get<3>(GetParam()));
+  save_out_data_ = get<4>(GetParam());
+
+  FILE* fp = fopen(in_filename_.c_str(), "rb");
+  assert(fp != NULL);
+
+  // Obtain file size.
+  fseek(fp, 0, SEEK_END);
+  loop_length_samples_ = ftell(fp) / sizeof(int16_t);
+  rewind(fp);
+
+  // Allocate memory to contain the whole file.
+  in_data_.reset(new int16_t[loop_length_samples_ +
+      input_length_sample_ * channels_]);
+
+  data_pointer_ = 0;
+
+  // Copy the file into the buffer.
+  ASSERT_EQ(fread(&in_data_[0], sizeof(int16_t), loop_length_samples_, fp),
+            loop_length_samples_);
+  fclose(fp);
+
+  // Add an extra block length of samples to the end of the array, starting
+  // over again from the beginning of the array. This is done to simplify
+  // the reading process when reading over the end of the loop.
+  memcpy(&in_data_[loop_length_samples_], &in_data_[0],
+         input_length_sample_ * channels_ * sizeof(int16_t));
+
+  max_bytes_ = input_length_sample_ * channels_ * sizeof(int16_t);
+  out_data_.reset(new int16_t[output_length_sample_ * channels_]);
+  bit_stream_.reset(new uint8_t[max_bytes_]);
+
+  if (save_out_data_) {
+    std::string out_filename =
+        ::testing::UnitTest::GetInstance()->current_test_info()->name();
+
+    // Erase '/'
+    size_t found;
+    while ((found = out_filename.find('/')) != std::string::npos)
+      out_filename.replace(found, 1, "_");
+
+    out_filename = test::OutputPath() + out_filename + ".pcm";
+
+    out_file_ = fopen(out_filename.c_str(), "wb");
+    assert(out_file_ != NULL);
+
+    printf("Output to be saved in %s.\n", out_filename.c_str());
+  }
+}
+
+void AudioCodecSpeedTest::TearDown() {
+  if (save_out_data_) {
+    fclose(out_file_);
+  }
+}
+
+void AudioCodecSpeedTest::EncodeDecode(size_t audio_duration_sec) {
+  size_t time_now_ms = 0;
+  float time_ms;
+
+  printf("Coding %d kHz-sampled %" PRIuS "-channel audio at %d bps ...\n",
+         input_sampling_khz_, channels_, bit_rate_);
+
+  while (time_now_ms < audio_duration_sec * 1000) {
+    // Encode & decode.
+    time_ms = EncodeABlock(&in_data_[data_pointer_], &bit_stream_[0],
+                           max_bytes_, &encoded_bytes_);
+    encoding_time_ms_ += time_ms;
+    time_ms = DecodeABlock(&bit_stream_[0], encoded_bytes_, &out_data_[0]);
+    decoding_time_ms_ += time_ms;
+    if (save_out_data_) {
+      fwrite(&out_data_[0], sizeof(int16_t),
+             output_length_sample_ * channels_, out_file_);
+    }
+    data_pointer_ = (data_pointer_ + input_length_sample_ * channels_) %
+        loop_length_samples_;
+    time_now_ms += block_duration_ms_;
+  }
+
+  printf("Encoding: %.2f%% real time,\nDecoding: %.2f%% real time.\n",
+         (encoding_time_ms_ / audio_duration_sec) / 10.0,
+         (decoding_time_ms_ / audio_duration_sec) / 10.0);
+}
+
+}  // namespace webrtc

modules/audio_coding/codecs/tools/audio_codec_speed_test.h

@@ -0,0 +1,92 @@
+/*
+ *  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 <memory>
+#include <string>
+
+#include "webrtc/test/gtest.h"
+#include "webrtc/typedefs.h"
+
+namespace webrtc {
+
+// Define coding parameter as
+// <channels, bit_rate, file_name, extension, if_save_output>.
+typedef std::tr1::tuple<size_t, 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,
+                             size_t max_bytes, size_t* 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, size_t 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.
+  size_t input_length_sample_;
+
+  // Expected output number of samples-per-channel in a frame.
+  size_t output_length_sample_;
+
+  std::unique_ptr<int16_t[]> in_data_;
+  std::unique_ptr<int16_t[]> out_data_;
+  size_t data_pointer_;
+  size_t loop_length_samples_;
+  std::unique_ptr<uint8_t[]> bit_stream_;
+
+  // Maximum number of bytes in output bitstream for a frame of audio.
+  size_t max_bytes_;
+
+  size_t encoded_bytes_;
+  float encoding_time_ms_;
+  float decoding_time_ms_;
+  FILE* out_file_;
+
+  size_t 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_