xref: /external/webrtc/webrtc/modules/audio_coding/acm2/acm_receive_test_oldapi.h

/*
 *  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_RECEIVE_TEST_OLDAPI_H_
#define WEBRTC_MODULES_AUDIO_CODING_ACM2_ACM_RECEIVE_TEST_OLDAPI_H_

#include <string>

#include "webrtc/base/constructormagic.h"
#include "webrtc/base/scoped_ptr.h"
#include "webrtc/system_wrappers/include/clock.h"

namespace webrtc {
class AudioCodingModule;
class AudioDecoder;
struct CodecInst;

namespace test {
class AudioSink;
class PacketSource;

class AcmReceiveTestOldApi {
 public:
  enum NumOutputChannels {
    kArbitraryChannels = 0,
    kMonoOutput = 1,
    kStereoOutput = 2
  };

  AcmReceiveTestOldApi(PacketSource* packet_source,
                       AudioSink* audio_sink,
                       int output_freq_hz,
                       NumOutputChannels exptected_output_channels);
  virtual ~AcmReceiveTestOldApi() {}

  // Registers the codecs with default parameters from ACM.
  void RegisterDefaultCodecs();

  // Registers codecs with payload types matching the pre-encoded NetEq test
  // files.
  void RegisterNetEqTestCodecs();

  int RegisterExternalReceiveCodec(int rtp_payload_type,
                                   AudioDecoder* external_decoder,
                                   int sample_rate_hz,
                                   int num_channels,
                                   const std::string& name);

  // Runs the test and returns true if successful.
  void Run();

 protected:
  // Method is called after each block of output audio is received from ACM.
  virtual void AfterGetAudio() {}

  SimulatedClock clock_;
  rtc::scoped_ptr<AudioCodingModule> acm_;
  PacketSource* packet_source_;
  AudioSink* audio_sink_;
  int output_freq_hz_;
  NumOutputChannels exptected_output_channels_;

  RTC_DISALLOW_COPY_AND_ASSIGN(AcmReceiveTestOldApi);
};

// This test toggles the output frequency every |toggle_period_ms|. The test
// starts with |output_freq_hz_1|. Except for the toggling, it does the same
// thing as AcmReceiveTestOldApi.
class AcmReceiveTestToggleOutputFreqOldApi : public AcmReceiveTestOldApi {
 public:
  AcmReceiveTestToggleOutputFreqOldApi(
      PacketSource* packet_source,
      AudioSink* audio_sink,
      int output_freq_hz_1,
      int output_freq_hz_2,
      int toggle_period_ms,
      NumOutputChannels exptected_output_channels);

 protected:
  void AfterGetAudio() override;

  const int output_freq_hz_1_;
  const int output_freq_hz_2_;
  const int toggle_period_ms_;
  int64_t last_toggle_time_ms_;
};

}  // namespace test
}  // namespace webrtc
#endif  // WEBRTC_MODULES_AUDIO_CODING_ACM2_ACM_RECEIVE_TEST_OLDAPI_H_