Move AudioOptions to its own header file and target.
It is part of our api. With the intention to later delete the inclusion of mediachannel.h from api/peerconnectioninterface.h, and eliminate circular dependencies. Bug: webrtc:7504 Change-Id: If44efd14d85675530e457760a1c4a1d338f931b7 Reviewed-on: https://webrtc-review.googlesource.com/41281 Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org> Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Commit-Queue: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21694}
This commit is contained in:
Niels Möller
Commit Bot
parent
e90316be3f
commit
a6fe261b97
13
api/BUILD.gn
13
api/BUILD.gn
@ -91,6 +91,7 @@ rtc_static_library("libjingle_peerconnection_api") {
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deps = [
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":array_view",
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":audio_options_api",
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":optional",
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":peerconnection_and_implicit_call_api",
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":rtc_stats_api",
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@ -197,6 +198,18 @@ rtc_source_set("audio_mixer_api") {
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]
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}
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rtc_source_set("audio_options_api") {
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visibility = [ "*" ]
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sources = [
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"audio_options.h",
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]
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deps = [
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":optional",
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"../rtc_base:rtc_base_approved",
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]
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}
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rtc_source_set("transport_api") {
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visibility = [ "*" ]
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sources = [
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206
api/audio_options.h
Normal file
206
api/audio_options.h
Normal file
@ -0,0 +1,206 @@
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/*
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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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#ifndef API_AUDIO_OPTIONS_H_
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#define API_AUDIO_OPTIONS_H_
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#include <string>
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#include "api/optional.h"
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#include "rtc_base/stringencode.h"
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namespace cricket {
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// Options that can be applied to a VoiceMediaChannel or a VoiceMediaEngine.
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// Used to be flags, but that makes it hard to selectively apply options.
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// We are moving all of the setting of options to structs like this,
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// but some things currently still use flags.
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struct AudioOptions {
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void 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(&typing_detection, change.typing_detection);
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SetFrom(&aecm_generate_comfort_noise, change.aecm_generate_comfort_noise);
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SetFrom(&experimental_agc, change.experimental_agc);
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SetFrom(&extended_filter_aec, change.extended_filter_aec);
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SetFrom(&delay_agnostic_aec, change.delay_agnostic_aec);
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SetFrom(&experimental_ns, change.experimental_ns);
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SetFrom(&intelligibility_enhancer, change.intelligibility_enhancer);
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SetFrom(&level_control, change.level_control);
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SetFrom(&residual_echo_detector, change.residual_echo_detector);
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SetFrom(&tx_agc_target_dbov, change.tx_agc_target_dbov);
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SetFrom(&tx_agc_digital_compression_gain,
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change.tx_agc_digital_compression_gain);
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SetFrom(&tx_agc_limiter, change.tx_agc_limiter);
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SetFrom(&combined_audio_video_bwe, change.combined_audio_video_bwe);
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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(&level_control_initial_peak_level_dbfs,
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change.level_control_initial_peak_level_dbfs);
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}
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bool 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 ==
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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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typing_detection == o.typing_detection &&
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aecm_generate_comfort_noise == o.aecm_generate_comfort_noise &&
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experimental_agc == o.experimental_agc &&
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extended_filter_aec == o.extended_filter_aec &&
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delay_agnostic_aec == o.delay_agnostic_aec &&
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experimental_ns == o.experimental_ns &&
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intelligibility_enhancer == o.intelligibility_enhancer &&
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level_control == o.level_control &&
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residual_echo_detector == o.residual_echo_detector &&
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tx_agc_target_dbov == o.tx_agc_target_dbov &&
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tx_agc_digital_compression_gain ==
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o.tx_agc_digital_compression_gain &&
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tx_agc_limiter == o.tx_agc_limiter &&
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combined_audio_video_bwe == o.combined_audio_video_bwe &&
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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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level_control_initial_peak_level_dbfs ==
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o.level_control_initial_peak_level_dbfs;
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}
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bool operator!=(const AudioOptions& o) const { return !(*this == o); }
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std::string ToString() const {
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std::ostringstream ost;
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ost << "AudioOptions {";
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ost << ToStringIfSet("aec", echo_cancellation);
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#if defined(WEBRTC_IOS)
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ost << ToStringIfSet("ios_force_software_aec_HACK",
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ios_force_software_aec_HACK);
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#endif
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ost << ToStringIfSet("agc", auto_gain_control);
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ost << ToStringIfSet("ns", noise_suppression);
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ost << ToStringIfSet("hf", highpass_filter);
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ost << ToStringIfSet("swap", stereo_swapping);
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ost << ToStringIfSet("audio_jitter_buffer_max_packets",
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audio_jitter_buffer_max_packets);
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ost << ToStringIfSet("audio_jitter_buffer_fast_accelerate",
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audio_jitter_buffer_fast_accelerate);
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ost << ToStringIfSet("typing", typing_detection);
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ost << ToStringIfSet("comfort_noise", aecm_generate_comfort_noise);
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ost << ToStringIfSet("experimental_agc", experimental_agc);
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ost << ToStringIfSet("extended_filter_aec", extended_filter_aec);
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ost << ToStringIfSet("delay_agnostic_aec", delay_agnostic_aec);
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ost << ToStringIfSet("experimental_ns", experimental_ns);
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ost << ToStringIfSet("intelligibility_enhancer", intelligibility_enhancer);
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ost << ToStringIfSet("level_control", level_control);
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ost << ToStringIfSet("level_control_initial_peak_level_dbfs",
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level_control_initial_peak_level_dbfs);
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ost << ToStringIfSet("residual_echo_detector", residual_echo_detector);
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ost << ToStringIfSet("tx_agc_target_dbov", tx_agc_target_dbov);
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ost << ToStringIfSet("tx_agc_digital_compression_gain",
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tx_agc_digital_compression_gain);
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ost << ToStringIfSet("tx_agc_limiter", tx_agc_limiter);
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ost << ToStringIfSet("combined_audio_video_bwe", combined_audio_video_bwe);
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ost << ToStringIfSet("audio_network_adaptor", audio_network_adaptor);
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// The adaptor config is a serialized proto buffer and therefore not human
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// readable. So we comment out the following line.
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// ost << ToStringIfSet("audio_network_adaptor_config",
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// audio_network_adaptor_config);
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ost << "}";
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return ost.str();
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}
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// Audio processing that attempts to filter away the output signal from
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// later inbound pickup.
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rtc::Optional<bool> echo_cancellation;
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#if defined(WEBRTC_IOS)
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// Forces software echo cancellation on iOS. This is a temporary workaround
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// (until Apple fixes the bug) for a device with non-functioning AEC. May
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// improve performance on that particular device, but will cause unpredictable
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// behavior in all other cases. See http://bugs.webrtc.org/8682.
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rtc::Optional<bool> ios_force_software_aec_HACK;
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#endif
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// Audio processing to adjust the sensitivity of the local mic dynamically.
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rtc::Optional<bool> auto_gain_control;
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// Audio processing to filter out background noise.
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rtc::Optional<bool> noise_suppression;
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// Audio processing to remove background noise of lower frequencies.
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rtc::Optional<bool> highpass_filter;
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// Audio processing to swap the left and right channels.
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rtc::Optional<bool> stereo_swapping;
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// Audio receiver jitter buffer (NetEq) max capacity in number of packets.
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rtc::Optional<int> audio_jitter_buffer_max_packets;
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// Audio receiver jitter buffer (NetEq) fast accelerate mode.
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rtc::Optional<bool> audio_jitter_buffer_fast_accelerate;
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// Audio processing to detect typing.
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rtc::Optional<bool> typing_detection;
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rtc::Optional<bool> aecm_generate_comfort_noise;
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rtc::Optional<bool> experimental_agc;
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rtc::Optional<bool> extended_filter_aec;
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rtc::Optional<bool> delay_agnostic_aec;
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rtc::Optional<bool> experimental_ns;
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rtc::Optional<bool> intelligibility_enhancer;
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rtc::Optional<bool> level_control;
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// Specifies an optional initialization value for the level controller.
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rtc::Optional<float> level_control_initial_peak_level_dbfs;
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// Note that tx_agc_* only applies to non-experimental AGC.
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rtc::Optional<bool> residual_echo_detector;
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rtc::Optional<uint16_t> tx_agc_target_dbov;
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rtc::Optional<uint16_t> tx_agc_digital_compression_gain;
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rtc::Optional<bool> tx_agc_limiter;
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// Enable combined audio+bandwidth BWE.
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// TODO(pthatcher): This flag is set from the
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// "googCombinedAudioVideoBwe", but not used anywhere. So delete it,
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// and check if any other AudioOptions members are unused.
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rtc::Optional<bool> combined_audio_video_bwe;
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// Enable audio network adaptor.
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rtc::Optional<bool> audio_network_adaptor;
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// Config string for audio network adaptor.
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rtc::Optional<std::string> audio_network_adaptor_config;
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private:
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template <class T>
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static std::string ToStringIfSet(const char* key,
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const rtc::Optional<T>& val) {
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std::string str;
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if (val) {
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str = key;
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str += ": ";
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str += val ? rtc::ToString(*val) : "";
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str += ", ";
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}
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return str;
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}
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template <typename T>
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static void SetFrom(rtc::Optional<T>* s, const rtc::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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};
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} // namespace cricket
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#endif // API_AUDIO_OPTIONS_H_
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@ -78,6 +78,7 @@
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#include "api/audio_codecs/audio_decoder_factory.h"
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#include "api/audio_codecs/audio_encoder_factory.h"
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#include "api/audio_options.h"
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#include "api/datachannelinterface.h"
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#include "api/dtmfsenderinterface.h"
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#include "api/jsep.h"
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@ -60,6 +60,7 @@ rtc_static_library("rtc_media_base") {
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defines = []
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libs = []
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deps = [
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"../api:audio_options_api",
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"../rtc_base:checks",
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"../rtc_base:sanitizer",
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"../rtc_base:stringutils",
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@ -18,6 +18,7 @@
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#include <vector>
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#include "api/audio_codecs/audio_encoder.h"
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#include "api/audio_options.h"
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#include "api/optional.h"
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#include "api/rtpparameters.h"
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#include "api/rtpreceiverinterface.h"
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@ -152,176 +153,6 @@ struct MediaConfig {
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bool operator!=(const MediaConfig& o) const { return !(*this == o); }
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};
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// Options that can be applied to a VoiceMediaChannel or a VoiceMediaEngine.
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// Used to be flags, but that makes it hard to selectively apply options.
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// We are moving all of the setting of options to structs like this,
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// but some things currently still use flags.
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struct AudioOptions {
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void 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(&typing_detection, change.typing_detection);
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SetFrom(&aecm_generate_comfort_noise, change.aecm_generate_comfort_noise);
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SetFrom(&experimental_agc, change.experimental_agc);
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SetFrom(&extended_filter_aec, change.extended_filter_aec);
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SetFrom(&delay_agnostic_aec, change.delay_agnostic_aec);
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SetFrom(&experimental_ns, change.experimental_ns);
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SetFrom(&intelligibility_enhancer, change.intelligibility_enhancer);
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SetFrom(&level_control, change.level_control);
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SetFrom(&residual_echo_detector, change.residual_echo_detector);
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SetFrom(&tx_agc_target_dbov, change.tx_agc_target_dbov);
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SetFrom(&tx_agc_digital_compression_gain,
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change.tx_agc_digital_compression_gain);
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SetFrom(&tx_agc_limiter, change.tx_agc_limiter);
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SetFrom(&combined_audio_video_bwe, change.combined_audio_video_bwe);
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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(&level_control_initial_peak_level_dbfs,
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change.level_control_initial_peak_level_dbfs);
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}
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bool 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 ==
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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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typing_detection == o.typing_detection &&
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aecm_generate_comfort_noise == o.aecm_generate_comfort_noise &&
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experimental_agc == o.experimental_agc &&
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extended_filter_aec == o.extended_filter_aec &&
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delay_agnostic_aec == o.delay_agnostic_aec &&
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experimental_ns == o.experimental_ns &&
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intelligibility_enhancer == o.intelligibility_enhancer &&
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level_control == o.level_control &&
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residual_echo_detector == o.residual_echo_detector &&
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tx_agc_target_dbov == o.tx_agc_target_dbov &&
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tx_agc_digital_compression_gain ==
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o.tx_agc_digital_compression_gain &&
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tx_agc_limiter == o.tx_agc_limiter &&
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combined_audio_video_bwe == o.combined_audio_video_bwe &&
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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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level_control_initial_peak_level_dbfs ==
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o.level_control_initial_peak_level_dbfs;
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}
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bool operator!=(const AudioOptions& o) const { return !(*this == o); }
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std::string ToString() const {
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std::ostringstream ost;
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ost << "AudioOptions {";
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ost << ToStringIfSet("aec", echo_cancellation);
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#if defined(WEBRTC_IOS)
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ost << ToStringIfSet("ios_force_software_aec_HACK",
|
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ios_force_software_aec_HACK);
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#endif
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ost << ToStringIfSet("agc", auto_gain_control);
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ost << ToStringIfSet("ns", noise_suppression);
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ost << ToStringIfSet("hf", highpass_filter);
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ost << ToStringIfSet("swap", stereo_swapping);
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ost << ToStringIfSet("audio_jitter_buffer_max_packets",
|
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audio_jitter_buffer_max_packets);
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ost << ToStringIfSet("audio_jitter_buffer_fast_accelerate",
|
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audio_jitter_buffer_fast_accelerate);
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ost << ToStringIfSet("typing", typing_detection);
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ost << ToStringIfSet("comfort_noise", aecm_generate_comfort_noise);
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ost << ToStringIfSet("experimental_agc", experimental_agc);
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ost << ToStringIfSet("extended_filter_aec", extended_filter_aec);
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ost << ToStringIfSet("delay_agnostic_aec", delay_agnostic_aec);
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ost << ToStringIfSet("experimental_ns", experimental_ns);
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ost << ToStringIfSet("intelligibility_enhancer", intelligibility_enhancer);
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ost << ToStringIfSet("level_control", level_control);
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ost << ToStringIfSet("level_control_initial_peak_level_dbfs",
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level_control_initial_peak_level_dbfs);
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ost << ToStringIfSet("residual_echo_detector", residual_echo_detector);
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ost << ToStringIfSet("tx_agc_target_dbov", tx_agc_target_dbov);
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ost << ToStringIfSet("tx_agc_digital_compression_gain",
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tx_agc_digital_compression_gain);
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ost << ToStringIfSet("tx_agc_limiter", tx_agc_limiter);
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ost << ToStringIfSet("combined_audio_video_bwe", combined_audio_video_bwe);
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||||
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 <typename T>
|
||||
static void SetFrom(rtc::Optional<T>* s, const rtc::Optional<T>& o) {
|
||||
if (o) {
|
||||
*s = o;
|
||||
}
|
||||
}
|
||||
};
|
||||
|
||||
// Options that can be applied to a VideoMediaChannel or a VideoMediaEngine.
|
||||
// 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,
|
||||
|
||||
Loading…
x
Reference in New Issue
Block a user