Remove legacy (unused) config param: jitter_buffer_enable_rtx_handling

Bug: none
Change-Id: I14164546950cc63c37e54544cdc80bfd4eddf211
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/262962
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Jakob Ivarsson‎ <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36955}

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);

api/audio_options.h

@@ -58,8 +58,6 @@ struct RTC_EXPORT AudioOptions {
   absl::optional<bool> audio_jitter_buffer_fast_accelerate;
   // Audio receiver jitter buffer (NetEq) minimum target delay in milliseconds.
   absl::optional<int> audio_jitter_buffer_min_delay_ms;
-  // Audio receiver jitter buffer (NetEq) should handle retransmitted packets.
-  absl::optional<bool> 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,

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.

audio/audio_receive_stream.cc

@@ -81,10 +81,9 @@ std::unique_ptr<voe::ChannelReceiveInterface> 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
 

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<AudioDecoderFactory> decoder_factory,
       absl::optional<AudioCodecPairId> 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<AudioDecoderFactory> decoder_factory,
     absl::optional<AudioCodecPairId> codec_pair_id,
@@ -1123,7 +1121,6 @@ std::unique_ptr<ChannelReceiveInterface> 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<AudioDecoderFactory> decoder_factory,
     absl::optional<AudioCodecPairId> codec_pair_id,
@@ -1134,9 +1131,8 @@ std::unique_ptr<ChannelReceiveInterface> 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

audio/channel_receive.h

@@ -182,7 +182,6 @@ std::unique_ptr<ChannelReceiveInterface> 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<AudioDecoderFactory> decoder_factory,
     absl::optional<AudioCodecPairId> codec_pair_id,

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

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<webrtc::FrameDecryptorInterface> frame_decryptor,
     const webrtc::CryptoOptions& crypto_options,
     rtc::scoped_refptr<webrtc::FrameTransformerInterface> 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_)));

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_;
 };

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 ==

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<rtc::RTCCertificate> certificate;
   if (!configuration.certificates.empty()) {