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}diff --git a/api/BUILD.gn b/api/BUILD.gn
index e3af7bb..59efbf8 100644
--- a/api/BUILD.gn
+++ b/api/BUILD.gn
@@ -91,6 +91,7 @@
deps = [
":array_view",
+ ":audio_options_api",
":optional",
":peerconnection_and_implicit_call_api",
":rtc_stats_api",
@@ -197,6 +198,18 @@
]
}
+rtc_source_set("audio_options_api") {
+ visibility = [ "*" ]
+ sources = [
+ "audio_options.h",
+ ]
+
+ deps = [
+ ":optional",
+ "../rtc_base:rtc_base_approved",
+ ]
+}
+
rtc_source_set("transport_api") {
visibility = [ "*" ]
sources = [
diff --git a/api/audio_options.h b/api/audio_options.h
new file mode 100644
index 0000000..d62e1f8
--- /dev/null
+++ b/api/audio_options.h
@@ -0,0 +1,206 @@
+/*
+ * 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_
diff --git a/api/peerconnectioninterface.h b/api/peerconnectioninterface.h
index f6fae58..0f8804a 100644
--- a/api/peerconnectioninterface.h
+++ b/api/peerconnectioninterface.h
@@ -78,6 +78,7 @@
#include "api/audio_codecs/audio_decoder_factory.h"
#include "api/audio_codecs/audio_encoder_factory.h"
+#include "api/audio_options.h"
#include "api/datachannelinterface.h"
#include "api/dtmfsenderinterface.h"
#include "api/jsep.h"