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/acm2/acm_send_test.h

@@ -0,0 +1,88 @@
+/*
+ *  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_ACM2_ACM_SEND_TEST_H_
+#define WEBRTC_MODULES_AUDIO_CODING_ACM2_ACM_SEND_TEST_H_
+
+#include <memory>
+#include <vector>
+
+#include "webrtc/modules/audio_coding/include/audio_coding_module.h"
+#include "webrtc/modules/audio_coding/neteq/tools/packet_source.h"
+#include "webrtc/rtc_base/constructormagic.h"
+#include "webrtc/system_wrappers/include/clock.h"
+
+namespace webrtc {
+class AudioEncoder;
+
+namespace test {
+class InputAudioFile;
+class Packet;
+
+class AcmSendTestOldApi : public AudioPacketizationCallback,
+                          public PacketSource {
+ public:
+  AcmSendTestOldApi(InputAudioFile* audio_source,
+                    int source_rate_hz,
+                    int test_duration_ms);
+  ~AcmSendTestOldApi() override;
+
+  // Registers the send codec. Returns true on success, false otherwise.
+  bool RegisterCodec(const char* payload_name,
+                     int sampling_freq_hz,
+                     int channels,
+                     int payload_type,
+                     int frame_size_samples);
+
+  // Registers an external send codec. Returns true on success, false otherwise.
+  bool RegisterExternalCodec(AudioEncoder* external_speech_encoder);
+
+  // Inherited from PacketSource.
+  std::unique_ptr<Packet> NextPacket() override;
+
+  // Inherited from AudioPacketizationCallback.
+  int32_t SendData(FrameType frame_type,
+                   uint8_t payload_type,
+                   uint32_t timestamp,
+                   const uint8_t* payload_data,
+                   size_t payload_len_bytes,
+                   const RTPFragmentationHeader* fragmentation) override;
+
+  AudioCodingModule* acm() { return acm_.get(); }
+
+ private:
+  static const int kBlockSizeMs = 10;
+
+  // Creates a Packet object from the last packet produced by ACM (and received
+  // through the SendData method as a callback).
+  std::unique_ptr<Packet> CreatePacket();
+
+  SimulatedClock clock_;
+  std::unique_ptr<AudioCodingModule> acm_;
+  InputAudioFile* audio_source_;
+  int source_rate_hz_;
+  const size_t input_block_size_samples_;
+  AudioFrame input_frame_;
+  bool codec_registered_;
+  int test_duration_ms_;
+  // The following member variables are set whenever SendData() is called.
+  FrameType frame_type_;
+  int payload_type_;
+  uint32_t timestamp_;
+  uint16_t sequence_number_;
+  std::vector<uint8_t> last_payload_vec_;
+  bool data_to_send_;
+
+  RTC_DISALLOW_COPY_AND_ASSIGN(AcmSendTestOldApi);
+};
+
+}  // namespace test
+}  // namespace webrtc
+#endif  // WEBRTC_MODULES_AUDIO_CODING_ACM2_ACM_SEND_TEST_H_