Removing AudioCoding class, a.k.a the new ACM API

webrtc/modules/audio_coding/main/acm2/acm_send_test.h

Index: webrtc/modules/audio_coding/main/acm2/acm_send_test.h
diff --git a/webrtc/modules/audio_coding/main/acm2/acm_send_test.h b/webrtc/modules/audio_coding/main/acm2/acm_send_test.h
deleted file mode 100644
index b0d26ba63b2010eda380889a11f3c3852f2809d4..0000000000000000000000000000000000000000
--- a/webrtc/modules/audio_coding/main/acm2/acm_send_test.h
+++ /dev/null
@@ -1,82 +0,0 @@
-/*
- *  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_MAIN_ACM2_ACM_SEND_TEST_H_
-#define WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_SEND_TEST_H_
-
-#include <vector>
-
-#include "webrtc/base/constructormagic.h"
-#include "webrtc/base/scoped_ptr.h"
-#include "webrtc/modules/audio_coding/main/interface/audio_coding_module.h"
-#include "webrtc/modules/audio_coding/neteq/tools/packet_source.h"
-#include "webrtc/system_wrappers/interface/clock.h"
-
-namespace webrtc {
-
-namespace test {
-class InputAudioFile;
-class Packet;
-
-class AcmSendTest : public AudioPacketizationCallback, public PacketSource {
- public:
-  AcmSendTest(InputAudioFile* audio_source,
-              int source_rate_hz,
-              int test_duration_ms);
-  virtual ~AcmSendTest() {}
-
-  // Registers the send codec. Returns true on success, false otherwise.
-  bool RegisterCodec(int codec_type,
-                     int channels,
-                     int payload_type,
-                     int frame_size_samples);
-
-  // Returns the next encoded packet. Returns NULL if the test duration was
-  // exceeded. Ownership of the packet is handed over to the caller.
-  // Inherited from PacketSource.
-  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;
-
- 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). Ownership of the new Packet
-  // object is transferred to the caller.
-  Packet* CreatePacket();
-
-  SimulatedClock clock_;
-  rtc::scoped_ptr<AudioCoding> 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_;
-
-  RTC_DISALLOW_COPY_AND_ASSIGN(AcmSendTest);
-};
-
-}  // namespace test
-}  // namespace webrtc
-#endif  // WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_SEND_TEST_H_