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/opus/opus_speed_test.cc

@@ -0,0 +1,121 @@
+/*
+ *  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/opus/opus_interface.h"
+#include "webrtc/modules/audio_coding/codecs/tools/audio_codec_speed_test.h"
+
+using ::std::string;
+
+namespace webrtc {
+
+static const int kOpusBlockDurationMs = 20;
+static const int kOpusSamplingKhz = 48;
+
+class OpusSpeedTest : public AudioCodecSpeedTest {
+ protected:
+  OpusSpeedTest();
+  void SetUp() override;
+  void TearDown() override;
+  float EncodeABlock(int16_t* in_data, uint8_t* bit_stream,
+                     size_t max_bytes, size_t* encoded_bytes) override;
+  float DecodeABlock(const uint8_t* bit_stream, size_t encoded_bytes,
+                     int16_t* out_data) override;
+  WebRtcOpusEncInst* opus_encoder_;
+  WebRtcOpusDecInst* opus_decoder_;
+};
+
+OpusSpeedTest::OpusSpeedTest()
+    : AudioCodecSpeedTest(kOpusBlockDurationMs,
+                          kOpusSamplingKhz,
+                          kOpusSamplingKhz),
+      opus_encoder_(NULL),
+      opus_decoder_(NULL) {
+}
+
+void OpusSpeedTest::SetUp() {
+  AudioCodecSpeedTest::SetUp();
+  // If channels_ == 1, use Opus VOIP mode, otherwise, audio mode.
+  int app = channels_ == 1 ? 0 : 1;
+  /* Create encoder memory. */
+  EXPECT_EQ(0, WebRtcOpus_EncoderCreate(&opus_encoder_, channels_, app));
+  EXPECT_EQ(0, WebRtcOpus_DecoderCreate(&opus_decoder_, channels_));
+  /* Set bitrate. */
+  EXPECT_EQ(0, WebRtcOpus_SetBitRate(opus_encoder_, bit_rate_));
+}
+
+void OpusSpeedTest::TearDown() {
+  AudioCodecSpeedTest::TearDown();
+  /* Free memory. */
+  EXPECT_EQ(0, WebRtcOpus_EncoderFree(opus_encoder_));
+  EXPECT_EQ(0, WebRtcOpus_DecoderFree(opus_decoder_));
+}
+
+float OpusSpeedTest::EncodeABlock(int16_t* in_data, uint8_t* bit_stream,
+                                  size_t max_bytes, size_t* encoded_bytes) {
+  clock_t clocks = clock();
+  int value = WebRtcOpus_Encode(opus_encoder_, in_data,
+                                input_length_sample_, max_bytes,
+                                bit_stream);
+  clocks = clock() - clocks;
+  EXPECT_GT(value, 0);
+  *encoded_bytes = static_cast<size_t>(value);
+  return 1000.0 * clocks / CLOCKS_PER_SEC;
+}
+
+float OpusSpeedTest::DecodeABlock(const uint8_t* bit_stream,
+                                  size_t encoded_bytes, int16_t* out_data) {
+  int value;
+  int16_t audio_type;
+  clock_t clocks = clock();
+  value = WebRtcOpus_Decode(opus_decoder_, bit_stream, encoded_bytes, out_data,
+                            &audio_type);
+  clocks = clock() - clocks;
+  EXPECT_EQ(output_length_sample_, static_cast<size_t>(value));
+  return 1000.0 * clocks / CLOCKS_PER_SEC;
+}
+
+#define ADD_TEST(complexity) \
+TEST_P(OpusSpeedTest, OpusSetComplexityTest##complexity) { \
+  /* Test audio length in second. */ \
+  size_t kDurationSec = 400; \
+  /* Set complexity. */ \
+  printf("Setting complexity to %d ...\n", complexity); \
+  EXPECT_EQ(0, WebRtcOpus_SetComplexity(opus_encoder_, complexity)); \
+  EncodeDecode(kDurationSec); \
+}
+
+ADD_TEST(10);
+ADD_TEST(9);
+ADD_TEST(8);
+ADD_TEST(7);
+ADD_TEST(6);
+ADD_TEST(5);
+ADD_TEST(4);
+ADD_TEST(3);
+ADD_TEST(2);
+ADD_TEST(1);
+ADD_TEST(0);
+
+// List all test cases: (channel, bit rat, filename, extension).
+const coding_param param_set[] =
+    {::std::tr1::make_tuple(1, 64000,
+                            string("audio_coding/speech_mono_32_48kHz"),
+                            string("pcm"), true),
+     ::std::tr1::make_tuple(1, 32000,
+                            string("audio_coding/speech_mono_32_48kHz"),
+                            string("pcm"), true),
+     ::std::tr1::make_tuple(2, 64000,
+                            string("audio_coding/music_stereo_48kHz"),
+                            string("pcm"), true)};
+
+INSTANTIATE_TEST_CASE_P(AllTest, OpusSpeedTest,
+                        ::testing::ValuesIn(param_set));
+
+}  // namespace webrtc