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/*
* Copyright (c) 2018 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.
*/
#include "api/audio_options.h"
#include "api/array_view.h"
#include "rtc_base/strings/string_builder.h"
namespace cricket {
namespace {
template <class T>
void ToStringIfSet(rtc::SimpleStringBuilder* result,
const char* key,
const absl::optional<T>& val) {
if (val) {
(*result) << key << ": " << *val << ", ";
}
}
template <typename T>
void SetFrom(absl::optional<T>* s, const absl::optional<T>& o) {
if (o) {
*s = o;
}
}
} // namespace
AudioOptions::AudioOptions() = default;
AudioOptions::~AudioOptions() = default;
void AudioOptions::SetAll(const AudioOptions& change) {
SetFrom(&echo_cancellation, change.echo_cancellation);
#if defined(WEBRTC_IOS)
SetFrom(&ios_force_software_aec_HACK, change.ios_force_software_aec_HACK);
#endif
SetFrom(&auto_gain_control, change.auto_gain_control);
SetFrom(&noise_suppression, change.noise_suppression);
SetFrom(&highpass_filter, change.highpass_filter);
SetFrom(&stereo_swapping, change.stereo_swapping);
SetFrom(&audio_jitter_buffer_max_packets,
change.audio_jitter_buffer_max_packets);
SetFrom(&audio_jitter_buffer_fast_accelerate,
change.audio_jitter_buffer_fast_accelerate);
SetFrom(&audio_jitter_buffer_min_delay_ms,
change.audio_jitter_buffer_min_delay_ms);
SetFrom(&combined_audio_video_bwe, change.combined_audio_video_bwe);
SetFrom(&audio_network_adaptor, change.audio_network_adaptor);
SetFrom(&audio_network_adaptor_config, change.audio_network_adaptor_config);
SetFrom(&init_recording_on_send, change.init_recording_on_send);
}
bool AudioOptions::operator==(const AudioOptions& o) const {
return echo_cancellation == o.echo_cancellation &&
#if defined(WEBRTC_IOS)
ios_force_software_aec_HACK == o.ios_force_software_aec_HACK &&
#endif
auto_gain_control == o.auto_gain_control &&
noise_suppression == o.noise_suppression &&
highpass_filter == o.highpass_filter &&
stereo_swapping == o.stereo_swapping &&
audio_jitter_buffer_max_packets == o.audio_jitter_buffer_max_packets &&
audio_jitter_buffer_fast_accelerate ==
o.audio_jitter_buffer_fast_accelerate &&
audio_jitter_buffer_min_delay_ms ==
o.audio_jitter_buffer_min_delay_ms &&
combined_audio_video_bwe == o.combined_audio_video_bwe &&
audio_network_adaptor == o.audio_network_adaptor &&
audio_network_adaptor_config == o.audio_network_adaptor_config &&
init_recording_on_send == o.init_recording_on_send;
}
std::string AudioOptions::ToString() const {
char buffer[1024];
rtc::SimpleStringBuilder result(buffer);
result << "AudioOptions {";
ToStringIfSet(&result, "aec", echo_cancellation);
#if defined(WEBRTC_IOS)
ToStringIfSet(&result, "ios_force_software_aec_HACK",
ios_force_software_aec_HACK);
#endif
ToStringIfSet(&result, "agc", auto_gain_control);
ToStringIfSet(&result, "ns", noise_suppression);
ToStringIfSet(&result, "hf", highpass_filter);
ToStringIfSet(&result, "swap", stereo_swapping);
ToStringIfSet(&result, "audio_jitter_buffer_max_packets",
audio_jitter_buffer_max_packets);
ToStringIfSet(&result, "audio_jitter_buffer_fast_accelerate",
audio_jitter_buffer_fast_accelerate);
ToStringIfSet(&result, "audio_jitter_buffer_min_delay_ms",
audio_jitter_buffer_min_delay_ms);
ToStringIfSet(&result, "combined_audio_video_bwe", combined_audio_video_bwe);
ToStringIfSet(&result, "audio_network_adaptor", audio_network_adaptor);
ToStringIfSet(&result, "init_recording_on_send", init_recording_on_send);
result << "}";
return result.str();
}
} // namespace cricket
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