1 files changed, 124 insertions, 0 deletions
diff --git a/chromium/third_party/webrtc/modules/audio_coding/codecs/tools/audio_codec_speed_test.cc b/chromium/third_party/webrtc/modules/audio_coding/codecs/tools/audio_codec_speed_test.cc
new file mode 100644
index 00000000000..c7cafdff9ba
--- /dev/null
+++ b/chromium/third_party/webrtc/modules/audio_coding/codecs/tools/audio_codec_speed_test.cc
@@ -0,0 +1,124 @@
+/*
+ *  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 "testing/gtest/include/gtest/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_(block_duration_ms_ * input_sampling_khz_),
+      output_length_sample_(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 %d-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