blob: d46411864e7d8b19ac7a0cc551429474c27bfa92 [file] [log] [blame]
Paulina Hensman11b34f42018-04-09 14:24:52 +02001/*
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
Danil Chapovalov21652332018-08-31 10:29:07 +020013#include "rtc_base/strings/string_builder.h"
14
Paulina Hensman11b34f42018-04-09 14:24:52 +020015namespace cricket {
Danil Chapovalov21652332018-08-31 10:29:07 +020016namespace {
17
18template <class T>
19void ToStringIfSet(rtc::SimpleStringBuilder* result,
20 const char* key,
21 const absl::optional<T>& val) {
22 if (val) {
23 (*result) << key << ": " << *val << ", ";
24 }
25}
26
27template <typename T>
28void SetFrom(absl::optional<T>* s, const absl::optional<T>& o) {
29 if (o) {
30 *s = o;
31 }
32}
33
34} // namespace
Paulina Hensman11b34f42018-04-09 14:24:52 +020035
36AudioOptions::AudioOptions() = default;
37AudioOptions::~AudioOptions() = default;
38
Danil Chapovalov21652332018-08-31 10:29:07 +020039void AudioOptions::SetAll(const AudioOptions& change) {
40 SetFrom(&echo_cancellation, change.echo_cancellation);
41#if defined(WEBRTC_IOS)
42 SetFrom(&ios_force_software_aec_HACK, change.ios_force_software_aec_HACK);
43#endif
44 SetFrom(&auto_gain_control, change.auto_gain_control);
45 SetFrom(&noise_suppression, change.noise_suppression);
46 SetFrom(&highpass_filter, change.highpass_filter);
47 SetFrom(&stereo_swapping, change.stereo_swapping);
48 SetFrom(&audio_jitter_buffer_max_packets,
49 change.audio_jitter_buffer_max_packets);
50 SetFrom(&audio_jitter_buffer_fast_accelerate,
51 change.audio_jitter_buffer_fast_accelerate);
52 SetFrom(&typing_detection, change.typing_detection);
Danil Chapovalov21652332018-08-31 10:29:07 +020053 SetFrom(&experimental_agc, change.experimental_agc);
54 SetFrom(&extended_filter_aec, change.extended_filter_aec);
55 SetFrom(&delay_agnostic_aec, change.delay_agnostic_aec);
56 SetFrom(&experimental_ns, change.experimental_ns);
57 SetFrom(&residual_echo_detector, change.residual_echo_detector);
58 SetFrom(&tx_agc_target_dbov, change.tx_agc_target_dbov);
59 SetFrom(&tx_agc_digital_compression_gain,
60 change.tx_agc_digital_compression_gain);
61 SetFrom(&tx_agc_limiter, change.tx_agc_limiter);
62 SetFrom(&combined_audio_video_bwe, change.combined_audio_video_bwe);
63 SetFrom(&audio_network_adaptor, change.audio_network_adaptor);
64 SetFrom(&audio_network_adaptor_config, change.audio_network_adaptor_config);
65}
66
67bool AudioOptions::operator==(const AudioOptions& o) const {
68 return echo_cancellation == o.echo_cancellation &&
69#if defined(WEBRTC_IOS)
70 ios_force_software_aec_HACK == o.ios_force_software_aec_HACK &&
71#endif
72 auto_gain_control == o.auto_gain_control &&
73 noise_suppression == o.noise_suppression &&
74 highpass_filter == o.highpass_filter &&
75 stereo_swapping == o.stereo_swapping &&
76 audio_jitter_buffer_max_packets == o.audio_jitter_buffer_max_packets &&
77 audio_jitter_buffer_fast_accelerate ==
78 o.audio_jitter_buffer_fast_accelerate &&
79 typing_detection == o.typing_detection &&
Danil Chapovalov21652332018-08-31 10:29:07 +020080 experimental_agc == o.experimental_agc &&
81 extended_filter_aec == o.extended_filter_aec &&
82 delay_agnostic_aec == o.delay_agnostic_aec &&
83 experimental_ns == o.experimental_ns &&
84 residual_echo_detector == o.residual_echo_detector &&
85 tx_agc_target_dbov == o.tx_agc_target_dbov &&
86 tx_agc_digital_compression_gain == o.tx_agc_digital_compression_gain &&
87 tx_agc_limiter == o.tx_agc_limiter &&
88 combined_audio_video_bwe == o.combined_audio_video_bwe &&
89 audio_network_adaptor == o.audio_network_adaptor &&
90 audio_network_adaptor_config == o.audio_network_adaptor_config;
91}
92
93std::string AudioOptions::ToString() const {
94 char buffer[1024];
95 rtc::SimpleStringBuilder result(buffer);
96 result << "AudioOptions {";
97 ToStringIfSet(&result, "aec", echo_cancellation);
98#if defined(WEBRTC_IOS)
99 ToStringIfSet(&result, "ios_force_software_aec_HACK",
100 ios_force_software_aec_HACK);
101#endif
102 ToStringIfSet(&result, "agc", auto_gain_control);
103 ToStringIfSet(&result, "ns", noise_suppression);
104 ToStringIfSet(&result, "hf", highpass_filter);
105 ToStringIfSet(&result, "swap", stereo_swapping);
106 ToStringIfSet(&result, "audio_jitter_buffer_max_packets",
107 audio_jitter_buffer_max_packets);
108 ToStringIfSet(&result, "audio_jitter_buffer_fast_accelerate",
109 audio_jitter_buffer_fast_accelerate);
110 ToStringIfSet(&result, "typing", typing_detection);
Danil Chapovalov21652332018-08-31 10:29:07 +0200111 ToStringIfSet(&result, "experimental_agc", experimental_agc);
112 ToStringIfSet(&result, "extended_filter_aec", extended_filter_aec);
113 ToStringIfSet(&result, "delay_agnostic_aec", delay_agnostic_aec);
114 ToStringIfSet(&result, "experimental_ns", experimental_ns);
115 ToStringIfSet(&result, "residual_echo_detector", residual_echo_detector);
116 ToStringIfSet(&result, "tx_agc_target_dbov", tx_agc_target_dbov);
117 ToStringIfSet(&result, "tx_agc_digital_compression_gain",
118 tx_agc_digital_compression_gain);
119 ToStringIfSet(&result, "tx_agc_limiter", tx_agc_limiter);
120 ToStringIfSet(&result, "combined_audio_video_bwe", combined_audio_video_bwe);
121 ToStringIfSet(&result, "audio_network_adaptor", audio_network_adaptor);
122 result << "}";
123 return result.str();
124}
125
Paulina Hensman11b34f42018-04-09 14:24:52 +0200126} // namespace cricket