audio_coding: remove "main" directory

webrtc/modules/audio_coding/main/test/TestStereo.h

-/*
- *  Copyright (c) 2012 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_TEST_TESTSTEREO_H_
-#define WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_TESTSTEREO_H_
-
-#include <math.h>
-
-#include "webrtc/base/scoped_ptr.h"
-#include "webrtc/modules/audio_coding/main/test/ACMTest.h"
-#include "webrtc/modules/audio_coding/main/test/Channel.h"
-#include "webrtc/modules/audio_coding/main/test/PCMFile.h"
-
-#define PCMA_AND_PCMU
-
-namespace webrtc {
-
-enum StereoMonoMode {
-  kNotSet,
-  kMono,
-  kStereo
-};
-
-class TestPackStereo : public AudioPacketizationCallback {
- public:
-  TestPackStereo();
-  ~TestPackStereo();
-
-  void RegisterReceiverACM(AudioCodingModule* acm);
-
-  int32_t SendData(const FrameType frame_type,
-                   const uint8_t payload_type,
-                   const uint32_t timestamp,
-                   const uint8_t* payload_data,
-                   const size_t payload_size,
-                   const RTPFragmentationHeader* fragmentation) override;
-
-  uint16_t payload_size();
-  uint32_t timestamp_diff();
-  void reset_payload_size();
-  void set_codec_mode(StereoMonoMode mode);
-  void set_lost_packet(bool lost);
-
- private:
-  AudioCodingModule* receiver_acm_;
-  int16_t seq_no_;
-  uint32_t timestamp_diff_;
-  uint32_t last_in_timestamp_;
-  uint64_t total_bytes_;
-  int payload_size_;
-  StereoMonoMode codec_mode_;
-  // Simulate packet losses
-  bool lost_packet_;
-};
-
-class TestStereo : public ACMTest {
- public:
-  explicit TestStereo(int test_mode);
-  ~TestStereo();
-
-  void Perform() override;
-
- private:
-  // The default value of '-1' indicates that the registration is based only on
-  // codec name and a sampling frequncy matching is not required. This is useful
-  // for codecs which support several sampling frequency.
-  void RegisterSendCodec(char side, char* codec_name, int32_t samp_freq_hz,
-                         int rate, int pack_size, int channels,
-                         int payload_type);
-
-  void Run(TestPackStereo* channel, int in_channels, int out_channels,
-           int percent_loss = 0);
-  void OpenOutFile(int16_t test_number);
-  void DisplaySendReceiveCodec();
-
-  int test_mode_;
-
-  rtc::scoped_ptr<AudioCodingModule> acm_a_;
-  rtc::scoped_ptr<AudioCodingModule> acm_b_;
-
-  TestPackStereo* channel_a2b_;
-
-  PCMFile* in_file_stereo_;
-  PCMFile* in_file_mono_;
-  PCMFile out_file_;
-  int16_t test_cntr_;
-  uint16_t pack_size_samp_;
-  uint16_t pack_size_bytes_;
-  int counter_;
-  char* send_codec_name_;
-
-  // Payload types for stereo codecs and CNG
-#ifdef WEBRTC_CODEC_G722
-  int g722_pltype_;
-#endif
-  int l16_8khz_pltype_;
-  int l16_16khz_pltype_;
-  int l16_32khz_pltype_;
-#ifdef PCMA_AND_PCMU
-  int pcma_pltype_;
-  int pcmu_pltype_;
-#endif
-#ifdef WEBRTC_CODEC_OPUS
-  int opus_pltype_;
-#endif
-};
-
-}  // namespace webrtc
-
-#endif  // WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_TESTSTEREO_H_