| /* | 
 |  *  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 "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(&typing_detection, change.typing_detection); | 
 |   SetFrom(&experimental_agc, change.experimental_agc); | 
 |   SetFrom(&extended_filter_aec, change.extended_filter_aec); | 
 |   SetFrom(&delay_agnostic_aec, change.delay_agnostic_aec); | 
 |   SetFrom(&experimental_ns, change.experimental_ns); | 
 |   SetFrom(&residual_echo_detector, change.residual_echo_detector); | 
 |   SetFrom(&tx_agc_target_dbov, change.tx_agc_target_dbov); | 
 |   SetFrom(&tx_agc_digital_compression_gain, | 
 |           change.tx_agc_digital_compression_gain); | 
 |   SetFrom(&tx_agc_limiter, change.tx_agc_limiter); | 
 |   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); | 
 | } | 
 |  | 
 | 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 && | 
 |          typing_detection == o.typing_detection && | 
 |          experimental_agc == o.experimental_agc && | 
 |          extended_filter_aec == o.extended_filter_aec && | 
 |          delay_agnostic_aec == o.delay_agnostic_aec && | 
 |          experimental_ns == o.experimental_ns && | 
 |          residual_echo_detector == o.residual_echo_detector && | 
 |          tx_agc_target_dbov == o.tx_agc_target_dbov && | 
 |          tx_agc_digital_compression_gain == o.tx_agc_digital_compression_gain && | 
 |          tx_agc_limiter == o.tx_agc_limiter && | 
 |          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; | 
 | } | 
 |  | 
 | 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, "typing", typing_detection); | 
 |   ToStringIfSet(&result, "experimental_agc", experimental_agc); | 
 |   ToStringIfSet(&result, "extended_filter_aec", extended_filter_aec); | 
 |   ToStringIfSet(&result, "delay_agnostic_aec", delay_agnostic_aec); | 
 |   ToStringIfSet(&result, "experimental_ns", experimental_ns); | 
 |   ToStringIfSet(&result, "residual_echo_detector", residual_echo_detector); | 
 |   ToStringIfSet(&result, "tx_agc_target_dbov", tx_agc_target_dbov); | 
 |   ToStringIfSet(&result, "tx_agc_digital_compression_gain", | 
 |                 tx_agc_digital_compression_gain); | 
 |   ToStringIfSet(&result, "tx_agc_limiter", tx_agc_limiter); | 
 |   ToStringIfSet(&result, "combined_audio_video_bwe", combined_audio_video_bwe); | 
 |   ToStringIfSet(&result, "audio_network_adaptor", audio_network_adaptor); | 
 |   result << "}"; | 
 |   return result.str(); | 
 | } | 
 |  | 
 | }  // namespace cricket |