diff --git a/api/audio_options.cc b/api/audio_options.cc index c301395f94..658515062c 100644 --- a/api/audio_options.cc +++ b/api/audio_options.cc @@ -52,8 +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(&audio_jitter_buffer_enable_rtx_handling, - change.audio_jitter_buffer_enable_rtx_handling); 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); @@ -74,8 +72,6 @@ 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 && - audio_jitter_buffer_enable_rtx_handling == - o.audio_jitter_buffer_enable_rtx_handling && 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 && @@ -101,8 +97,6 @@ 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, "audio_jitter_buffer_enable_rtx_handling", - audio_jitter_buffer_enable_rtx_handling); 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); diff --git a/api/audio_options.h b/api/audio_options.h index 1e4c820597..39ba3886ea 100644 --- a/api/audio_options.h +++ b/api/audio_options.h @@ -58,8 +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; - // Audio receiver jitter buffer (NetEq) should handle retransmitted packets. - absl::optional audio_jitter_buffer_enable_rtx_handling; // Enable combined audio+bandwidth BWE. // TODO(pthatcher): This flag is set from the // "googCombinedAudioVideoBwe", but not used anywhere. So delete it, diff --git a/api/peer_connection_interface.h b/api/peer_connection_interface.h index c326799edb..d42521cc46 100644 --- a/api/peer_connection_interface.h +++ b/api/peer_connection_interface.h @@ -491,10 +491,6 @@ class RTC_EXPORT PeerConnectionInterface : public rtc::RefCountInterface { // The minimum delay in milliseconds for the audio jitter buffer. int audio_jitter_buffer_min_delay_ms = 0; - // Whether the audio jitter buffer adapts the delay to retransmitted - // packets. - bool audio_jitter_buffer_enable_rtx_handling = false; - // Timeout in milliseconds before an ICE candidate pair is considered to be // "not receiving", after which a lower priority candidate pair may be // selected. diff --git a/audio/audio_receive_stream.cc b/audio/audio_receive_stream.cc index d12edf37c3..bc7dddde5c 100644 --- a/audio/audio_receive_stream.cc +++ b/audio/audio_receive_stream.cc @@ -81,10 +81,9 @@ std::unique_ptr CreateChannelReceive( config.rtcp_send_transport, event_log, config.rtp.local_ssrc, config.rtp.remote_ssrc, config.jitter_buffer_max_packets, config.jitter_buffer_fast_accelerate, config.jitter_buffer_min_delay_ms, - config.jitter_buffer_enable_rtx_handling, config.enable_non_sender_rtt, - config.decoder_factory, config.codec_pair_id, - std::move(config.frame_decryptor), config.crypto_options, - std::move(config.frame_transformer)); + config.enable_non_sender_rtt, config.decoder_factory, + config.codec_pair_id, std::move(config.frame_decryptor), + config.crypto_options, std::move(config.frame_transformer)); } } // namespace diff --git a/audio/channel_receive.cc b/audio/channel_receive.cc index d00a9a9469..fcf3dc1dd2 100644 --- a/audio/channel_receive.cc +++ b/audio/channel_receive.cc @@ -94,7 +94,6 @@ class ChannelReceive : public ChannelReceiveInterface, size_t jitter_buffer_max_packets, bool jitter_buffer_fast_playout, int jitter_buffer_min_delay_ms, - bool jitter_buffer_enable_rtx_handling, bool enable_non_sender_rtt, rtc::scoped_refptr decoder_factory, absl::optional codec_pair_id, @@ -523,7 +522,6 @@ ChannelReceive::ChannelReceive( size_t jitter_buffer_max_packets, bool jitter_buffer_fast_playout, int jitter_buffer_min_delay_ms, - bool jitter_buffer_enable_rtx_handling, bool enable_non_sender_rtt, rtc::scoped_refptr decoder_factory, absl::optional codec_pair_id, @@ -1123,7 +1121,6 @@ std::unique_ptr CreateChannelReceive( size_t jitter_buffer_max_packets, bool jitter_buffer_fast_playout, int jitter_buffer_min_delay_ms, - bool jitter_buffer_enable_rtx_handling, bool enable_non_sender_rtt, rtc::scoped_refptr decoder_factory, absl::optional codec_pair_id, @@ -1134,9 +1131,8 @@ std::unique_ptr CreateChannelReceive( clock, neteq_factory, audio_device_module, rtcp_send_transport, rtc_event_log, local_ssrc, remote_ssrc, jitter_buffer_max_packets, jitter_buffer_fast_playout, jitter_buffer_min_delay_ms, - jitter_buffer_enable_rtx_handling, enable_non_sender_rtt, decoder_factory, - codec_pair_id, std::move(frame_decryptor), crypto_options, - std::move(frame_transformer)); + enable_non_sender_rtt, decoder_factory, codec_pair_id, + std::move(frame_decryptor), crypto_options, std::move(frame_transformer)); } } // namespace voe diff --git a/audio/channel_receive.h b/audio/channel_receive.h index d811e87719..1c3b192a62 100644 --- a/audio/channel_receive.h +++ b/audio/channel_receive.h @@ -182,7 +182,6 @@ std::unique_ptr CreateChannelReceive( size_t jitter_buffer_max_packets, bool jitter_buffer_fast_playout, int jitter_buffer_min_delay_ms, - bool jitter_buffer_enable_rtx_handling, bool enable_non_sender_rtt, rtc::scoped_refptr decoder_factory, absl::optional codec_pair_id, diff --git a/call/audio_receive_stream.h b/call/audio_receive_stream.h index 94b52a2885..2a6ed6079e 100644 --- a/call/audio_receive_stream.h +++ b/call/audio_receive_stream.h @@ -127,7 +127,6 @@ class AudioReceiveStream : public MediaReceiveStreamInterface { size_t jitter_buffer_max_packets = 200; bool jitter_buffer_fast_accelerate = false; int jitter_buffer_min_delay_ms = 0; - bool jitter_buffer_enable_rtx_handling = false; // Identifier for an A/V synchronization group. Empty string to disable. // TODO(pbos): Synchronize streams in a sync group, not just one video diff --git a/media/engine/webrtc_voice_engine.cc b/media/engine/webrtc_voice_engine.cc index 82a30ffbf2..710823ac8f 100644 --- a/media/engine/webrtc_voice_engine.cc +++ b/media/engine/webrtc_voice_engine.cc @@ -260,7 +260,6 @@ webrtc::AudioReceiveStream::Config BuildReceiveStreamConfig( size_t jitter_buffer_max_packets, bool jitter_buffer_fast_accelerate, int jitter_buffer_min_delay_ms, - bool jitter_buffer_enable_rtx_handling, rtc::scoped_refptr frame_decryptor, const webrtc::CryptoOptions& crypto_options, rtc::scoped_refptr frame_transformer) { @@ -281,7 +280,6 @@ webrtc::AudioReceiveStream::Config BuildReceiveStreamConfig( config.jitter_buffer_max_packets = jitter_buffer_max_packets; config.jitter_buffer_fast_accelerate = jitter_buffer_fast_accelerate; config.jitter_buffer_min_delay_ms = jitter_buffer_min_delay_ms; - config.jitter_buffer_enable_rtx_handling = jitter_buffer_enable_rtx_handling; config.frame_decryptor = std::move(frame_decryptor); config.crypto_options = crypto_options; config.frame_transformer = std::move(frame_transformer); @@ -400,7 +398,6 @@ void WebRtcVoiceEngine::Init() { options.audio_jitter_buffer_max_packets = 200; options.audio_jitter_buffer_fast_accelerate = false; options.audio_jitter_buffer_min_delay_ms = 0; - options.audio_jitter_buffer_enable_rtx_handling = false; bool error = ApplyOptions(options); RTC_DCHECK(error); } @@ -548,12 +545,6 @@ bool WebRtcVoiceEngine::ApplyOptions(const AudioOptions& options_in) { audio_jitter_buffer_min_delay_ms_ = *options.audio_jitter_buffer_min_delay_ms; } - if (options.audio_jitter_buffer_enable_rtx_handling) { - RTC_LOG(LS_INFO) << "NetEq handle reordered packets? " - << *options.audio_jitter_buffer_enable_rtx_handling; - audio_jitter_buffer_enable_rtx_handling_ = - *options.audio_jitter_buffer_enable_rtx_handling; - } webrtc::AudioProcessing* ap = apm(); if (!ap) { @@ -1958,9 +1949,8 @@ bool WebRtcVoiceMediaChannel::AddRecvStream(const StreamParams& sp) { recv_rtp_extensions_, this, engine()->decoder_factory_, decoder_map_, codec_pair_id_, engine()->audio_jitter_buffer_max_packets_, engine()->audio_jitter_buffer_fast_accelerate_, - engine()->audio_jitter_buffer_min_delay_ms_, - engine()->audio_jitter_buffer_enable_rtx_handling_, - unsignaled_frame_decryptor_, crypto_options_, nullptr); + engine()->audio_jitter_buffer_min_delay_ms_, unsignaled_frame_decryptor_, + crypto_options_, nullptr); recv_streams_.insert(std::make_pair( ssrc, new WebRtcAudioReceiveStream(std::move(config), call_))); diff --git a/media/engine/webrtc_voice_engine.h b/media/engine/webrtc_voice_engine.h index f15b9f5882..e28a12a869 100644 --- a/media/engine/webrtc_voice_engine.h +++ b/media/engine/webrtc_voice_engine.h @@ -128,7 +128,6 @@ class WebRtcVoiceEngine final : public VoiceEngineInterface { size_t audio_jitter_buffer_max_packets_ = 200; bool audio_jitter_buffer_fast_accelerate_ = false; int audio_jitter_buffer_min_delay_ms_ = 0; - bool audio_jitter_buffer_enable_rtx_handling_ = false; const bool minimized_remsampling_on_mobile_trial_enabled_; }; diff --git a/pc/peer_connection.cc b/pc/peer_connection.cc index f914fc8365..6ab7527720 100644 --- a/pc/peer_connection.cc +++ b/pc/peer_connection.cc @@ -315,7 +315,6 @@ bool PeerConnectionInterface::RTCConfiguration::operator==( int audio_jitter_buffer_max_packets; bool audio_jitter_buffer_fast_accelerate; int audio_jitter_buffer_min_delay_ms; - bool audio_jitter_buffer_enable_rtx_handling; int ice_connection_receiving_timeout; int ice_backup_candidate_pair_ping_interval; ContinualGatheringPolicy continual_gathering_policy; @@ -362,8 +361,6 @@ bool PeerConnectionInterface::RTCConfiguration::operator==( o.audio_jitter_buffer_fast_accelerate && audio_jitter_buffer_min_delay_ms == o.audio_jitter_buffer_min_delay_ms && - audio_jitter_buffer_enable_rtx_handling == - o.audio_jitter_buffer_enable_rtx_handling && ice_connection_receiving_timeout == o.ice_connection_receiving_timeout && ice_backup_candidate_pair_ping_interval == diff --git a/pc/sdp_offer_answer.cc b/pc/sdp_offer_answer.cc index b1afd5c8f4..fc52a4a99b 100644 --- a/pc/sdp_offer_answer.cc +++ b/pc/sdp_offer_answer.cc @@ -1211,9 +1211,6 @@ void SdpOfferAnswerHandler::Initialize( audio_options_.audio_jitter_buffer_min_delay_ms = configuration.audio_jitter_buffer_min_delay_ms; - audio_options_.audio_jitter_buffer_enable_rtx_handling = - configuration.audio_jitter_buffer_enable_rtx_handling; - // Obtain a certificate from RTCConfiguration if any were provided (optional). rtc::scoped_refptr certificate; if (!configuration.certificates.empty()) {