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/media/base/mediachannel.h b/media/base/mediachannel.h
index ccec53f..c0b5d2b 100644
--- a/media/base/mediachannel.h
+++ b/media/base/mediachannel.h
@@ -18,6 +18,7 @@
#include <vector>
#include "api/audio_codecs/audio_encoder.h"
+#include "api/audio_options.h"
#include "api/optional.h"
#include "api/rtpparameters.h"
#include "api/rtpreceiverinterface.h"
@@ -152,176 +153,6 @@
bool operator!=(const MediaConfig& o) const { return !(*this == o); }
};
-// 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 <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,