mirror of
https://github.com/mollyim/webrtc.git
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87e74f9fb7
Bug: None Change-Id: Ie539351f98b7a0ebb5f08e0df5c5759a2bcb5588 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/306520 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Per Kjellander <perkj@webrtc.org> Reviewed-by: Stefan Holmer <stefan@webrtc.org> Commit-Queue: Yury Yarashevich <yura.yaroshevich@gmail.com> Cr-Commit-Position: refs/heads/main@{#40160}
104 lines
3.8 KiB
C++
104 lines
3.8 KiB
C++
/*
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* Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#include "api/audio_options.h"
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#include "api/array_view.h"
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#include "rtc_base/strings/string_builder.h"
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namespace cricket {
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namespace {
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template <class T>
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void ToStringIfSet(rtc::SimpleStringBuilder* result,
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const char* key,
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const absl::optional<T>& val) {
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if (val) {
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(*result) << key << ": " << *val << ", ";
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}
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}
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template <typename T>
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void SetFrom(absl::optional<T>* s, const absl::optional<T>& o) {
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if (o) {
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*s = o;
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}
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}
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} // namespace
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AudioOptions::AudioOptions() = default;
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AudioOptions::~AudioOptions() = default;
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void AudioOptions::SetAll(const AudioOptions& change) {
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SetFrom(&echo_cancellation, change.echo_cancellation);
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#if defined(WEBRTC_IOS)
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SetFrom(&ios_force_software_aec_HACK, change.ios_force_software_aec_HACK);
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#endif
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SetFrom(&auto_gain_control, change.auto_gain_control);
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SetFrom(&noise_suppression, change.noise_suppression);
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SetFrom(&highpass_filter, change.highpass_filter);
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SetFrom(&stereo_swapping, change.stereo_swapping);
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SetFrom(&audio_jitter_buffer_max_packets,
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change.audio_jitter_buffer_max_packets);
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SetFrom(&audio_jitter_buffer_fast_accelerate,
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change.audio_jitter_buffer_fast_accelerate);
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SetFrom(&audio_jitter_buffer_min_delay_ms,
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change.audio_jitter_buffer_min_delay_ms);
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SetFrom(&audio_network_adaptor, change.audio_network_adaptor);
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SetFrom(&audio_network_adaptor_config, change.audio_network_adaptor_config);
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SetFrom(&init_recording_on_send, change.init_recording_on_send);
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}
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bool AudioOptions::operator==(const AudioOptions& o) const {
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return echo_cancellation == o.echo_cancellation &&
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#if defined(WEBRTC_IOS)
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ios_force_software_aec_HACK == o.ios_force_software_aec_HACK &&
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#endif
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auto_gain_control == o.auto_gain_control &&
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noise_suppression == o.noise_suppression &&
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highpass_filter == o.highpass_filter &&
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stereo_swapping == o.stereo_swapping &&
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audio_jitter_buffer_max_packets == o.audio_jitter_buffer_max_packets &&
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audio_jitter_buffer_fast_accelerate ==
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o.audio_jitter_buffer_fast_accelerate &&
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audio_jitter_buffer_min_delay_ms ==
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o.audio_jitter_buffer_min_delay_ms &&
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audio_network_adaptor == o.audio_network_adaptor &&
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audio_network_adaptor_config == o.audio_network_adaptor_config &&
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init_recording_on_send == o.init_recording_on_send;
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}
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std::string AudioOptions::ToString() const {
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char buffer[1024];
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rtc::SimpleStringBuilder result(buffer);
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result << "AudioOptions {";
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ToStringIfSet(&result, "aec", echo_cancellation);
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#if defined(WEBRTC_IOS)
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ToStringIfSet(&result, "ios_force_software_aec_HACK",
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ios_force_software_aec_HACK);
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#endif
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ToStringIfSet(&result, "agc", auto_gain_control);
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ToStringIfSet(&result, "ns", noise_suppression);
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ToStringIfSet(&result, "hf", highpass_filter);
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ToStringIfSet(&result, "swap", stereo_swapping);
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ToStringIfSet(&result, "audio_jitter_buffer_max_packets",
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audio_jitter_buffer_max_packets);
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ToStringIfSet(&result, "audio_jitter_buffer_fast_accelerate",
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audio_jitter_buffer_fast_accelerate);
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ToStringIfSet(&result, "audio_jitter_buffer_min_delay_ms",
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audio_jitter_buffer_min_delay_ms);
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ToStringIfSet(&result, "audio_network_adaptor", audio_network_adaptor);
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ToStringIfSet(&result, "init_recording_on_send", init_recording_on_send);
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result << "}";
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return result.str();
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}
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} // namespace cricket
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