File: acm_receive_test.h

package info (click to toggle)
chromium-browser 70.0.3538.110-1~deb9u1
  • links: PTS, VCS
  • area: main
  • in suites: stretch
  • size: 1,619,476 kB
  • sloc: cpp: 13,024,755; ansic: 1,349,823; python: 916,672; xml: 314,489; java: 280,047; asm: 276,936; perl: 75,771; objc: 66,634; sh: 45,860; cs: 28,354; php: 11,064; makefile: 10,911; yacc: 9,109; tcl: 8,403; ruby: 4,065; lex: 1,779; pascal: 1,411; lisp: 1,055; awk: 41; jsp: 39; sed: 17; sql: 3
file content (97 lines) | stat: -rw-r--r-- 2,889 bytes parent folder | download
1
2
3
4
5
6
7
8
9
10
11
12
13
14
15
16
17
18
19
20
21
22
23
24
25
26
27
28
29
30
31
32
33
34
35
36
37
38
39
40
41
42
43
44
45
46
47
48
49
50
51
52
53
54
55
56
57
58
59
60
61
62
63
64
65
66
67
68
69
70
71
72
73
74
75
76
77
78
79
80
81
82
83
84
85
86
87
88
89
90
91
92
93
94
95
96
97
/*
 *  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 MODULES_AUDIO_CODING_ACM2_ACM_RECEIVE_TEST_H_
#define MODULES_AUDIO_CODING_ACM2_ACM_RECEIVE_TEST_H_

#include <stddef.h>  // for size_t
#include <memory>
#include <string>

#include "api/audio_codecs/audio_decoder_factory.h"
#include "rtc_base/constructormagic.h"
#include "rtc_base/scoped_ref_ptr.h"
#include "system_wrappers/include/clock.h"

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

namespace test {
class AudioSink;
class PacketSource;

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

  AcmReceiveTestOldApi(PacketSource* packet_source,
                       AudioSink* audio_sink,
                       int output_freq_hz,
                       NumOutputChannels exptected_output_channels,
                       rtc::scoped_refptr<AudioDecoderFactory> decoder_factory);
  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();

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

  AudioCodingModule* get_acm() { return acm_.get(); }

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

  SimulatedClock clock_;
  std::unique_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  // MODULES_AUDIO_CODING_ACM2_ACM_RECEIVE_TEST_H_