api/audio_options.h - src - Git at Google

 /*
  *  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.
  */

 #ifndef API_AUDIO_OPTIONS_H_
 #define API_AUDIO_OPTIONS_H_

 #include <string>

 #include "api/optional.h"
 #include "rtc_base/stringencode.h"

 namespace cricket {

 // Options that can be applied to a VoiceMediaChannel or a VoiceMediaEngine.
 // Used to be flags, but that makes it hard to selectively apply options.
 // We are moving all of the setting of options to structs like this,
 // but some things currently still use flags.
 struct AudioOptions {
   void 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(&aecm_generate_comfort_noise, change.aecm_generate_comfort_noise);
     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(&intelligibility_enhancer, change.intelligibility_enhancer);
     SetFrom(&level_control, change.level_control);
     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);
     SetFrom(&level_control_initial_peak_level_dbfs,
             change.level_control_initial_peak_level_dbfs);
   }

   bool 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 &&
            aecm_generate_comfort_noise == o.aecm_generate_comfort_noise &&
            experimental_agc == o.experimental_agc &&
            extended_filter_aec == o.extended_filter_aec &&
            delay_agnostic_aec == o.delay_agnostic_aec &&
            experimental_ns == o.experimental_ns &&
            intelligibility_enhancer == o.intelligibility_enhancer &&
            level_control == o.level_control &&
            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 &&
            level_control_initial_peak_level_dbfs ==
                o.level_control_initial_peak_level_dbfs;
   }
   bool operator!=(const AudioOptions& o) const { return !(*this == o); }

   std::string ToString() const {
     std::ostringstream ost;
     ost << "AudioOptions {";
     ost << ToStringIfSet("aec", echo_cancellation);
 #if defined(WEBRTC_IOS)
     ost << ToStringIfSet("ios_force_software_aec_HACK",
                          ios_force_software_aec_HACK);
 #endif
     ost << ToStringIfSet("agc", auto_gain_control);
     ost << ToStringIfSet("ns", noise_suppression);
     ost << ToStringIfSet("hf", highpass_filter);
     ost << ToStringIfSet("swap", stereo_swapping);
     ost << ToStringIfSet("audio_jitter_buffer_max_packets",
                          audio_jitter_buffer_max_packets);
     ost << ToStringIfSet("audio_jitter_buffer_fast_accelerate",
                          audio_jitter_buffer_fast_accelerate);
     ost << ToStringIfSet("typing", typing_detection);
     ost << ToStringIfSet("comfort_noise", aecm_generate_comfort_noise);
     ost << ToStringIfSet("experimental_agc", experimental_agc);
     ost << ToStringIfSet("extended_filter_aec", extended_filter_aec);
     ost << ToStringIfSet("delay_agnostic_aec", delay_agnostic_aec);
     ost << ToStringIfSet("experimental_ns", experimental_ns);
     ost << ToStringIfSet("intelligibility_enhancer", intelligibility_enhancer);
     ost << ToStringIfSet("level_control", level_control);
     ost << ToStringIfSet("level_control_initial_peak_level_dbfs",
                          level_control_initial_peak_level_dbfs);
     ost << ToStringIfSet("residual_echo_detector", residual_echo_detector);
     ost << ToStringIfSet("tx_agc_target_dbov", tx_agc_target_dbov);
     ost << ToStringIfSet("tx_agc_digital_compression_gain",
         tx_agc_digital_compression_gain);
     ost << ToStringIfSet("tx_agc_limiter", tx_agc_limiter);
     ost << ToStringIfSet("combined_audio_video_bwe", combined_audio_video_bwe);
     ost << ToStringIfSet("audio_network_adaptor", audio_network_adaptor);
     // The adaptor config is a serialized proto buffer and therefore not human
     // readable. So we comment out the following line.
     // ost << ToStringIfSet("audio_network_adaptor_config",
     //     audio_network_adaptor_config);
     ost << "}";
     return ost.str();
   }

   // Audio processing that attempts to filter away the output signal from
   // later inbound pickup.
   rtc::Optional<bool> echo_cancellation;
 #if defined(WEBRTC_IOS)
   // Forces software echo cancellation on iOS. This is a temporary workaround
   // (until Apple fixes the bug) for a device with non-functioning AEC. May
   // improve performance on that particular device, but will cause unpredictable
   // behavior in all other cases. See http://bugs.webrtc.org/8682.
   rtc::Optional<bool> ios_force_software_aec_HACK;
 #endif
   // Audio processing to adjust the sensitivity of the local mic dynamically.
   rtc::Optional<bool> auto_gain_control;
   // Audio processing to filter out background noise.
   rtc::Optional<bool> noise_suppression;
   // Audio processing to remove background noise of lower frequencies.
   rtc::Optional<bool> highpass_filter;
   // Audio processing to swap the left and right channels.
   rtc::Optional<bool> stereo_swapping;
   // Audio receiver jitter buffer (NetEq) max capacity in number of packets.
   rtc::Optional<int> audio_jitter_buffer_max_packets;
   // Audio receiver jitter buffer (NetEq) fast accelerate mode.
   rtc::Optional<bool> audio_jitter_buffer_fast_accelerate;
   // Audio processing to detect typing.
   rtc::Optional<bool> typing_detection;
   rtc::Optional<bool> aecm_generate_comfort_noise;
   rtc::Optional<bool> experimental_agc;
   rtc::Optional<bool> extended_filter_aec;
   rtc::Optional<bool> delay_agnostic_aec;
   rtc::Optional<bool> experimental_ns;
   rtc::Optional<bool> intelligibility_enhancer;
   rtc::Optional<bool> level_control;
   // Specifies an optional initialization value for the level controller.
   rtc::Optional<float> level_control_initial_peak_level_dbfs;
   // Note that tx_agc_* only applies to non-experimental AGC.
   rtc::Optional<bool> residual_echo_detector;
   rtc::Optional<uint16_t> tx_agc_target_dbov;
   rtc::Optional<uint16_t> tx_agc_digital_compression_gain;
   rtc::Optional<bool> tx_agc_limiter;
   // Enable combined audio+bandwidth BWE.
   // TODO(pthatcher): This flag is set from the
   // "googCombinedAudioVideoBwe", but not used anywhere. So delete it,
   // and check if any other AudioOptions members are unused.
   rtc::Optional<bool> combined_audio_video_bwe;
   // Enable audio network adaptor.
   rtc::Optional<bool> audio_network_adaptor;
   // Config string for audio network adaptor.
   rtc::Optional<std::string> audio_network_adaptor_config;

  private:
   template <class T>
   static std::string ToStringIfSet(const char* key,
                                    const rtc::Optional<T>& val) {
     std::string str;
     if (val) {
       str = key;
       str += ": ";
       str += val ? rtc::ToString(*val) : "";
       str += ", ";
     }
     return str;
   }

   template <typename T>
   static void SetFrom(rtc::Optional<T>* s, const rtc::Optional<T>& o) {
     if (o) {
       *s = o;
     }
   }
 };

 }  // namespace cricket

 #endif  // API_AUDIO_OPTIONS_H_
	/*
	* 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.
	*/

	#ifndef API_AUDIO_OPTIONS_H_
	#define API_AUDIO_OPTIONS_H_

	#include <string>

	#include "api/optional.h"
	#include "rtc_base/stringencode.h"

	namespace cricket {

	// Options that can be applied to a VoiceMediaChannel or a VoiceMediaEngine.
	// Used to be flags, but that makes it hard to selectively apply options.
	// We are moving all of the setting of options to structs like this,
	// but some things currently still use flags.
	struct AudioOptions {
	void 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(&aecm_generate_comfort_noise, change.aecm_generate_comfort_noise);
	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(&intelligibility_enhancer, change.intelligibility_enhancer);
	SetFrom(&level_control, change.level_control);
	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);
	SetFrom(&level_control_initial_peak_level_dbfs,
	change.level_control_initial_peak_level_dbfs);
	}

	bool 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 &&
	aecm_generate_comfort_noise == o.aecm_generate_comfort_noise &&
	experimental_agc == o.experimental_agc &&
	extended_filter_aec == o.extended_filter_aec &&
	delay_agnostic_aec == o.delay_agnostic_aec &&
	experimental_ns == o.experimental_ns &&
	intelligibility_enhancer == o.intelligibility_enhancer &&
	level_control == o.level_control &&
	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 &&
	level_control_initial_peak_level_dbfs ==
	o.level_control_initial_peak_level_dbfs;
	}
	bool operator!=(const AudioOptions& o) const { return !(*this == o); }

	std::string ToString() const {
	std::ostringstream ost;
	ost << "AudioOptions {";
	ost << ToStringIfSet("aec", echo_cancellation);
	#if defined(WEBRTC_IOS)
	ost << ToStringIfSet("ios_force_software_aec_HACK",
	ios_force_software_aec_HACK);
	#endif
	ost << ToStringIfSet("agc", auto_gain_control);
	ost << ToStringIfSet("ns", noise_suppression);
	ost << ToStringIfSet("hf", highpass_filter);
	ost << ToStringIfSet("swap", stereo_swapping);
	ost << ToStringIfSet("audio_jitter_buffer_max_packets",
	audio_jitter_buffer_max_packets);
	ost << ToStringIfSet("audio_jitter_buffer_fast_accelerate",
	audio_jitter_buffer_fast_accelerate);
	ost << ToStringIfSet("typing", typing_detection);
	ost << ToStringIfSet("comfort_noise", aecm_generate_comfort_noise);
	ost << ToStringIfSet("experimental_agc", experimental_agc);
	ost << ToStringIfSet("extended_filter_aec", extended_filter_aec);
	ost << ToStringIfSet("delay_agnostic_aec", delay_agnostic_aec);
	ost << ToStringIfSet("experimental_ns", experimental_ns);
	ost << ToStringIfSet("intelligibility_enhancer", intelligibility_enhancer);
	ost << ToStringIfSet("level_control", level_control);
	ost << ToStringIfSet("level_control_initial_peak_level_dbfs",
	level_control_initial_peak_level_dbfs);
	ost << ToStringIfSet("residual_echo_detector", residual_echo_detector);
	ost << ToStringIfSet("tx_agc_target_dbov", tx_agc_target_dbov);
	ost << ToStringIfSet("tx_agc_digital_compression_gain",
	tx_agc_digital_compression_gain);
	ost << ToStringIfSet("tx_agc_limiter", tx_agc_limiter);
	ost << ToStringIfSet("combined_audio_video_bwe", combined_audio_video_bwe);
	ost << ToStringIfSet("audio_network_adaptor", audio_network_adaptor);
	// The adaptor config is a serialized proto buffer and therefore not human
	// readable. So we comment out the following line.
	// ost << ToStringIfSet("audio_network_adaptor_config",
	// audio_network_adaptor_config);
	ost << "}";
	return ost.str();
	}

	// Audio processing that attempts to filter away the output signal from
	// later inbound pickup.
	rtc::Optional<bool> echo_cancellation;
	#if defined(WEBRTC_IOS)
	// Forces software echo cancellation on iOS. This is a temporary workaround
	// (until Apple fixes the bug) for a device with non-functioning AEC. May
	// improve performance on that particular device, but will cause unpredictable
	// behavior in all other cases. See http://bugs.webrtc.org/8682.
	rtc::Optional<bool> ios_force_software_aec_HACK;
	#endif
	// Audio processing to adjust the sensitivity of the local mic dynamically.
	rtc::Optional<bool> auto_gain_control;
	// Audio processing to filter out background noise.
	rtc::Optional<bool> noise_suppression;
	// Audio processing to remove background noise of lower frequencies.
	rtc::Optional<bool> highpass_filter;
	// Audio processing to swap the left and right channels.
	rtc::Optional<bool> stereo_swapping;
	// Audio receiver jitter buffer (NetEq) max capacity in number of packets.
	rtc::Optional<int> audio_jitter_buffer_max_packets;
	// Audio receiver jitter buffer (NetEq) fast accelerate mode.
	rtc::Optional<bool> audio_jitter_buffer_fast_accelerate;
	// Audio processing to detect typing.
	rtc::Optional<bool> typing_detection;
	rtc::Optional<bool> aecm_generate_comfort_noise;
	rtc::Optional<bool> experimental_agc;
	rtc::Optional<bool> extended_filter_aec;
	rtc::Optional<bool> delay_agnostic_aec;
	rtc::Optional<bool> experimental_ns;
	rtc::Optional<bool> intelligibility_enhancer;
	rtc::Optional<bool> level_control;
	// Specifies an optional initialization value for the level controller.
	rtc::Optional<float> level_control_initial_peak_level_dbfs;
	// Note that tx_agc_* only applies to non-experimental AGC.
	rtc::Optional<bool> residual_echo_detector;
	rtc::Optional<uint16_t> tx_agc_target_dbov;
	rtc::Optional<uint16_t> tx_agc_digital_compression_gain;
	rtc::Optional<bool> tx_agc_limiter;
	// Enable combined audio+bandwidth BWE.
	// TODO(pthatcher): This flag is set from the
	// "googCombinedAudioVideoBwe", but not used anywhere. So delete it,
	// and check if any other AudioOptions members are unused.
	rtc::Optional<bool> combined_audio_video_bwe;
	// Enable audio network adaptor.
	rtc::Optional<bool> audio_network_adaptor;
	// Config string for audio network adaptor.
	rtc::Optional<std::string> audio_network_adaptor_config;

	private:
	template <class T>
	static std::string ToStringIfSet(const char* key,
	const rtc::Optional<T>& val) {
	std::string str;
	if (val) {
	str = key;
	str += ": ";
	str += val ? rtc::ToString(*val) : "";
	str += ", ";
	}
	return str;
	}

	template <typename T>
	static void SetFrom(rtc::Optional<T>* s, const rtc::Optional<T>& o) {
	if (o) {
	*s = o;
	}
	}
	};

	} // namespace cricket

	#endif // API_AUDIO_OPTIONS_H_