1 /*
2 * Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
3 *
4 * Use of this source code is governed by a BSD-style license
5 * that can be found in the LICENSE file in the root of the source
6 * tree. An additional intellectual property rights grant can be found
7 * in the file PATENTS. All contributing project authors may
8 * be found in the AUTHORS file in the root of the source tree.
9 */
10
11 #include "api/audio_options.h"
12
13 #include "api/array_view.h"
14 #include "rtc_base/strings/string_builder.h"
15
16 namespace cricket {
17 namespace {
18
19 template <class T>
ToStringIfSet(rtc::SimpleStringBuilder * result,const char * key,const absl::optional<T> & val)20 void ToStringIfSet(rtc::SimpleStringBuilder* result,
21 const char* key,
22 const absl::optional<T>& val) {
23 if (val) {
24 (*result) << key << ": " << *val << ", ";
25 }
26 }
27
28 template <typename T>
SetFrom(absl::optional<T> * s,const absl::optional<T> & o)29 void SetFrom(absl::optional<T>* s, const absl::optional<T>& o) {
30 if (o) {
31 *s = o;
32 }
33 }
34
35 } // namespace
36
37 AudioOptions::AudioOptions() = default;
38 AudioOptions::~AudioOptions() = default;
39
SetAll(const AudioOptions & change)40 void AudioOptions::SetAll(const AudioOptions& change) {
41 SetFrom(&echo_cancellation, change.echo_cancellation);
42 #if defined(WEBRTC_IOS)
43 SetFrom(&ios_force_software_aec_HACK, change.ios_force_software_aec_HACK);
44 #endif
45 SetFrom(&auto_gain_control, change.auto_gain_control);
46 SetFrom(&noise_suppression, change.noise_suppression);
47 SetFrom(&highpass_filter, change.highpass_filter);
48 SetFrom(&stereo_swapping, change.stereo_swapping);
49 SetFrom(&audio_jitter_buffer_max_packets,
50 change.audio_jitter_buffer_max_packets);
51 SetFrom(&audio_jitter_buffer_fast_accelerate,
52 change.audio_jitter_buffer_fast_accelerate);
53 SetFrom(&audio_jitter_buffer_min_delay_ms,
54 change.audio_jitter_buffer_min_delay_ms);
55 SetFrom(&audio_jitter_buffer_enable_rtx_handling,
56 change.audio_jitter_buffer_enable_rtx_handling);
57 SetFrom(&typing_detection, change.typing_detection);
58 SetFrom(&experimental_agc, change.experimental_agc);
59 SetFrom(&experimental_ns, change.experimental_ns);
60 SetFrom(&residual_echo_detector, change.residual_echo_detector);
61 SetFrom(&tx_agc_target_dbov, change.tx_agc_target_dbov);
62 SetFrom(&tx_agc_digital_compression_gain,
63 change.tx_agc_digital_compression_gain);
64 SetFrom(&tx_agc_limiter, change.tx_agc_limiter);
65 SetFrom(&combined_audio_video_bwe, change.combined_audio_video_bwe);
66 SetFrom(&audio_network_adaptor, change.audio_network_adaptor);
67 SetFrom(&audio_network_adaptor_config, change.audio_network_adaptor_config);
68 }
69
operator ==(const AudioOptions & o) const70 bool AudioOptions::operator==(const AudioOptions& o) const {
71 return echo_cancellation == o.echo_cancellation &&
72 #if defined(WEBRTC_IOS)
73 ios_force_software_aec_HACK == o.ios_force_software_aec_HACK &&
74 #endif
75 auto_gain_control == o.auto_gain_control &&
76 noise_suppression == o.noise_suppression &&
77 highpass_filter == o.highpass_filter &&
78 stereo_swapping == o.stereo_swapping &&
79 audio_jitter_buffer_max_packets == o.audio_jitter_buffer_max_packets &&
80 audio_jitter_buffer_fast_accelerate ==
81 o.audio_jitter_buffer_fast_accelerate &&
82 audio_jitter_buffer_min_delay_ms ==
83 o.audio_jitter_buffer_min_delay_ms &&
84 audio_jitter_buffer_enable_rtx_handling ==
85 o.audio_jitter_buffer_enable_rtx_handling &&
86 typing_detection == o.typing_detection &&
87 experimental_agc == o.experimental_agc &&
88 experimental_ns == o.experimental_ns &&
89 residual_echo_detector == o.residual_echo_detector &&
90 tx_agc_target_dbov == o.tx_agc_target_dbov &&
91 tx_agc_digital_compression_gain == o.tx_agc_digital_compression_gain &&
92 tx_agc_limiter == o.tx_agc_limiter &&
93 combined_audio_video_bwe == o.combined_audio_video_bwe &&
94 audio_network_adaptor == o.audio_network_adaptor &&
95 audio_network_adaptor_config == o.audio_network_adaptor_config;
96 }
97
ToString() const98 std::string AudioOptions::ToString() const {
99 char buffer[1024];
100 rtc::SimpleStringBuilder result(buffer);
101 result << "AudioOptions {";
102 ToStringIfSet(&result, "aec", echo_cancellation);
103 #if defined(WEBRTC_IOS)
104 ToStringIfSet(&result, "ios_force_software_aec_HACK",
105 ios_force_software_aec_HACK);
106 #endif
107 ToStringIfSet(&result, "agc", auto_gain_control);
108 ToStringIfSet(&result, "ns", noise_suppression);
109 ToStringIfSet(&result, "hf", highpass_filter);
110 ToStringIfSet(&result, "swap", stereo_swapping);
111 ToStringIfSet(&result, "audio_jitter_buffer_max_packets",
112 audio_jitter_buffer_max_packets);
113 ToStringIfSet(&result, "audio_jitter_buffer_fast_accelerate",
114 audio_jitter_buffer_fast_accelerate);
115 ToStringIfSet(&result, "audio_jitter_buffer_min_delay_ms",
116 audio_jitter_buffer_min_delay_ms);
117 ToStringIfSet(&result, "audio_jitter_buffer_enable_rtx_handling",
118 audio_jitter_buffer_enable_rtx_handling);
119 ToStringIfSet(&result, "typing", typing_detection);
120 ToStringIfSet(&result, "experimental_agc", experimental_agc);
121 ToStringIfSet(&result, "experimental_ns", experimental_ns);
122 ToStringIfSet(&result, "residual_echo_detector", residual_echo_detector);
123 ToStringIfSet(&result, "tx_agc_target_dbov", tx_agc_target_dbov);
124 ToStringIfSet(&result, "tx_agc_digital_compression_gain",
125 tx_agc_digital_compression_gain);
126 ToStringIfSet(&result, "tx_agc_limiter", tx_agc_limiter);
127 ToStringIfSet(&result, "combined_audio_video_bwe", combined_audio_video_bwe);
128 ToStringIfSet(&result, "audio_network_adaptor", audio_network_adaptor);
129 result << "}";
130 return result.str();
131 }
132
133 } // namespace cricket
134