diff --git a/api/audio_options.cc b/api/audio_options.cc index 658515062c..a3f2b6e887 100644 --- a/api/audio_options.cc +++ b/api/audio_options.cc @@ -52,7 +52,6 @@ void AudioOptions::SetAll(const AudioOptions& change) { change.audio_jitter_buffer_fast_accelerate); SetFrom(&audio_jitter_buffer_min_delay_ms, change.audio_jitter_buffer_min_delay_ms); - 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(&init_recording_on_send, change.init_recording_on_send); @@ -72,7 +71,6 @@ bool AudioOptions::operator==(const AudioOptions& o) const { o.audio_jitter_buffer_fast_accelerate && audio_jitter_buffer_min_delay_ms == o.audio_jitter_buffer_min_delay_ms && - 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 && init_recording_on_send == o.init_recording_on_send; @@ -97,7 +95,6 @@ std::string AudioOptions::ToString() const { audio_jitter_buffer_fast_accelerate); ToStringIfSet(&result, "audio_jitter_buffer_min_delay_ms", audio_jitter_buffer_min_delay_ms); - ToStringIfSet(&result, "combined_audio_video_bwe", combined_audio_video_bwe); ToStringIfSet(&result, "audio_network_adaptor", audio_network_adaptor); ToStringIfSet(&result, "init_recording_on_send", init_recording_on_send); result << "}"; diff --git a/api/audio_options.h b/api/audio_options.h index 39ba3886ea..3ab3b3c98c 100644 --- a/api/audio_options.h +++ b/api/audio_options.h @@ -58,11 +58,6 @@ struct RTC_EXPORT AudioOptions { absl::optional audio_jitter_buffer_fast_accelerate; // Audio receiver jitter buffer (NetEq) minimum target delay in milliseconds. absl::optional audio_jitter_buffer_min_delay_ms; - // 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. - absl::optional combined_audio_video_bwe; // Enable audio network adaptor. // TODO(webrtc:11717): Remove this API in favor of adaptivePtime in // RtpEncodingParameters. diff --git a/api/peer_connection_interface.h b/api/peer_connection_interface.h index 6ce4650e5f..e80550cb07 100644 --- a/api/peer_connection_interface.h +++ b/api/peer_connection_interface.h @@ -448,9 +448,6 @@ class RTC_EXPORT PeerConnectionInterface : public rtc::RefCountInterface { // when switching from a static scene to one with motion. absl::optional screencast_min_bitrate; - // Use new combined audio/video bandwidth estimation? - absl::optional combined_audio_video_bwe; - #if defined(WEBRTC_FUCHSIA) // TODO(bugs.webrtc.org/11066): Remove entirely once Fuchsia does not use. // TODO(bugs.webrtc.org/9891) - Move to crypto_options diff --git a/pc/peer_connection.cc b/pc/peer_connection.cc index 59d38727dc..e0cef06714 100644 --- a/pc/peer_connection.cc +++ b/pc/peer_connection.cc @@ -303,7 +303,6 @@ bool PeerConnectionInterface::RTCConfiguration::operator==( int max_ipv6_networks; bool disable_link_local_networks; absl::optional screencast_min_bitrate; - absl::optional combined_audio_video_bwe; #if defined(WEBRTC_FUCHSIA) absl::optional enable_dtls_srtp; #endif @@ -372,7 +371,6 @@ bool PeerConnectionInterface::RTCConfiguration::operator==( max_ipv6_networks == o.max_ipv6_networks && disable_link_local_networks == o.disable_link_local_networks && screencast_min_bitrate == o.screencast_min_bitrate && - combined_audio_video_bwe == o.combined_audio_video_bwe && #if defined(WEBRTC_FUCHSIA) enable_dtls_srtp == o.enable_dtls_srtp && #endif diff --git a/pc/peer_connection_media_unittest.cc b/pc/peer_connection_media_unittest.cc index 56ff8ffa35..2213db6153 100644 --- a/pc/peer_connection_media_unittest.cc +++ b/pc/peer_connection_media_unittest.cc @@ -1348,21 +1348,6 @@ TEST_P(PeerConnectionMediaTest, SetRemoteDescriptionFailsWithDuplicateMids) { "Failed to set remote offer sdp: Duplicate a=mid value 'same'."); } -TEST_P(PeerConnectionMediaTest, - CombinedAudioVideoBweConfigPropagatedToMediaEngine) { - RTCConfiguration config; - config.combined_audio_video_bwe.emplace(true); - auto caller = CreatePeerConnectionWithAudioVideo(config); - - ASSERT_TRUE(caller->SetLocalDescription(caller->CreateOffer())); - - auto caller_voice = caller->media_engine()->GetVoiceSendChannel(0); - ASSERT_TRUE(caller_voice); - const cricket::AudioOptions& audio_options = caller_voice->options(); - EXPECT_EQ(config.combined_audio_video_bwe, - audio_options.combined_audio_video_bwe); -} - // Test that if a RED codec refers to another codec in its fmtp line, but that // codec's payload type was reassigned for some reason (either the remote // endpoint selected a different payload type or there was a conflict), the RED diff --git a/pc/sdp_offer_answer.cc b/pc/sdp_offer_answer.cc index 7f3db91a5c..c87b6ec641 100644 --- a/pc/sdp_offer_answer.cc +++ b/pc/sdp_offer_answer.cc @@ -1326,8 +1326,6 @@ void SdpOfferAnswerHandler::Initialize( // RTCConfiguration value (not available on Web). video_options_.screencast_min_bitrate_kbps = configuration.screencast_min_bitrate.value_or(100); - audio_options_.combined_audio_video_bwe = - configuration.combined_audio_video_bwe; audio_options_.audio_jitter_buffer_max_packets = configuration.audio_jitter_buffer_max_packets; diff --git a/sdk/android/api/org/webrtc/PeerConnection.java b/sdk/android/api/org/webrtc/PeerConnection.java index e334a3fd9d..5c87fe3ea3 100644 --- a/sdk/android/api/org/webrtc/PeerConnection.java +++ b/sdk/android/api/org/webrtc/PeerConnection.java @@ -528,7 +528,6 @@ public class PeerConnection { public boolean enableCpuOveruseDetection; public boolean suspendBelowMinBitrate; @Nullable public Integer screencastMinBitrate; - @Nullable public Boolean combinedAudioVideoBwe; // Use "Unknown" to represent no preference of adapter types, not the // preference of adapters of unknown types. public AdapterType networkPreference; @@ -607,7 +606,6 @@ public class PeerConnection { enableCpuOveruseDetection = true; suspendBelowMinBitrate = false; screencastMinBitrate = null; - combinedAudioVideoBwe = null; networkPreference = AdapterType.UNKNOWN; sdpSemantics = SdpSemantics.UNIFIED_PLAN; activeResetSrtpParams = false; @@ -788,12 +786,6 @@ public class PeerConnection { return screencastMinBitrate; } - @Nullable - @CalledByNative("RTCConfiguration") - Boolean getCombinedAudioVideoBwe() { - return combinedAudioVideoBwe; - } - @CalledByNative("RTCConfiguration") AdapterType getNetworkPreference() { return networkPreference; diff --git a/sdk/android/src/jni/pc/peer_connection.cc b/sdk/android/src/jni/pc/peer_connection.cc index 9983ae7df2..a063804ac8 100644 --- a/sdk/android/src/jni/pc/peer_connection.cc +++ b/sdk/android/src/jni/pc/peer_connection.cc @@ -260,8 +260,6 @@ void JavaToNativeRTCConfiguration( Java_RTCConfiguration_getSuspendBelowMinBitrate(jni, j_rtc_config); rtc_config->screencast_min_bitrate = JavaToNativeOptionalInt( jni, Java_RTCConfiguration_getScreencastMinBitrate(jni, j_rtc_config)); - rtc_config->combined_audio_video_bwe = JavaToNativeOptionalBool( - jni, Java_RTCConfiguration_getCombinedAudioVideoBwe(jni, j_rtc_config)); rtc_config->network_preference = JavaToNativeNetworkPreference(jni, j_network_preference); rtc_config->sdp_semantics = JavaToNativeSdpSemantics(jni, j_sdp_semantics); diff --git a/sdk/media_constraints.cc b/sdk/media_constraints.cc index bbb46edaae..88261e7530 100644 --- a/sdk/media_constraints.cc +++ b/sdk/media_constraints.cc @@ -117,8 +117,6 @@ const char MediaConstraints::kUseRtpMux[] = "googUseRtpMUX"; const char MediaConstraints::kEnableDscp[] = "googDscp"; const char MediaConstraints::kEnableVideoSuspendBelowMinBitrate[] = "googSuspendBelowMinBitrate"; -const char MediaConstraints::kCombinedAudioVideoBwe[] = - "googCombinedAudioVideoBwe"; const char MediaConstraints::kScreencastMinBitrate[] = "googScreencastMinBitrate"; // TODO(ronghuawu): Remove once cpu overuse detection is stable. @@ -162,9 +160,6 @@ void CopyConstraintsIntoRtcConfiguration( ConstraintToOptional(constraints, MediaConstraints::kScreencastMinBitrate, &configuration->screencast_min_bitrate); - ConstraintToOptional(constraints, - MediaConstraints::kCombinedAudioVideoBwe, - &configuration->combined_audio_video_bwe); } void CopyConstraintsIntoAudioOptions(const MediaConstraints* constraints, diff --git a/sdk/media_constraints.h b/sdk/media_constraints.h index a428abdce0..5bd38c2d1a 100644 --- a/sdk/media_constraints.h +++ b/sdk/media_constraints.h @@ -89,9 +89,6 @@ class MediaConstraints { static const char kEnableIPv6[]; // googIPv6 // Temporary constraint to enable suspend below min bitrate feature. static const char kEnableVideoSuspendBelowMinBitrate[]; - // googSuspendBelowMinBitrate - // Constraint to enable combined audio+video bandwidth estimation. - static const char kCombinedAudioVideoBwe[]; // googCombinedAudioVideoBwe static const char kScreencastMinBitrate[]; // googScreencastMinBitrate static const char kCpuOveruseDetection[]; // googCpuOveruseDetection diff --git a/sdk/media_constraints_unittest.cc b/sdk/media_constraints_unittest.cc index 2d25da03e7..5e6b157e3a 100644 --- a/sdk/media_constraints_unittest.cc +++ b/sdk/media_constraints_unittest.cc @@ -23,7 +23,6 @@ bool Matches(const PeerConnectionInterface::RTCConfiguration& a, return a.audio_jitter_buffer_max_packets == b.audio_jitter_buffer_max_packets && a.screencast_min_bitrate == b.screencast_min_bitrate && - a.combined_audio_video_bwe == b.combined_audio_video_bwe && a.media_config == b.media_config; }