commit 9ece54fa7385e4b190bad48f97a4ec69eaad4459
author Per K <perkj@webrtc.org> Wed Jan 18 13:22:38 2023
committer WebRTC LUCI CQ <webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com> Wed Jan 18 14:06:33 2023
tree 338616e0fdecaf0deacbd6a394218807ad68c7cb
parent 444741e78d7b61b16e0984c15073877bac52914e

Delete unnecssary AudioReceiveStreamInterface::GetRtpExtensions

Bug: webrtc:7135, webrtc:14795
Change-Id: I0242a3600d4a156eae2315966e5e59e03be8aeab
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290998
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39139}

audio/audio_receive_stream.cc

diff --git a/audio/audio_receive_stream.cc b/audio/audio_receive_stream.cc
index 7a2037e..0bb1168 100644
--- a/audio/audio_receive_stream.cc
+++ b/audio/audio_receive_stream.cc
@@ -262,12 +262,6 @@
   config_.rtp.extensions = std::move(extensions);
 }
 
-const std::vector<RtpExtension>& AudioReceiveStreamImpl::GetRtpExtensions()
-    const {
-  RTC_DCHECK_RUN_ON(&worker_thread_checker_);
-  return config_.rtp.extensions;
-}
-
 RtpHeaderExtensionMap AudioReceiveStreamImpl::GetRtpExtensionMap() const {
   return RtpHeaderExtensionMap(config_.rtp.extensions);
 }

audio/audio_receive_stream.h

diff --git a/audio/audio_receive_stream.h b/audio/audio_receive_stream.h
index d9283ec..51514fb 100644
--- a/audio/audio_receive_stream.h
+++ b/audio/audio_receive_stream.h
@@ -95,7 +95,6 @@
   void SetFrameDecryptor(rtc::scoped_refptr<webrtc::FrameDecryptorInterface>
                              frame_decryptor) override;
   void SetRtpExtensions(std::vector<RtpExtension> extensions) override;
-  const std::vector<RtpExtension>& GetRtpExtensions() const override;
   RtpHeaderExtensionMap GetRtpExtensionMap() const override;
 
   webrtc::AudioReceiveStreamInterface::Stats GetStats(

call/audio_receive_stream.h

diff --git a/call/audio_receive_stream.h b/call/audio_receive_stream.h
index f383277..1228861 100644
--- a/call/audio_receive_stream.h
+++ b/call/audio_receive_stream.h
@@ -198,12 +198,6 @@
   // post initialization.
   virtual uint32_t remote_ssrc() const = 0;
 
-  // Access the currently set rtp extensions. Must be called on the packet
-  // delivery thread.
-  // TODO(tommi): This is currently only called from
-  // `WebRtcAudioReceiveStream::GetRtpParameters()`. See if we can remove it.
-  virtual const std::vector<RtpExtension>& GetRtpExtensions() const = 0;
-
  protected:
   virtual ~AudioReceiveStreamInterface() {}
 };

media/engine/fake_webrtc_call.cc

diff --git a/media/engine/fake_webrtc_call.cc b/media/engine/fake_webrtc_call.cc
index ef9224e..a20b826 100644
--- a/media/engine/fake_webrtc_call.cc
+++ b/media/engine/fake_webrtc_call.cc
@@ -135,11 +135,6 @@
   config_.rtp.extensions = std::move(extensions);
 }
 
-const std::vector<webrtc::RtpExtension>&
-FakeAudioReceiveStream::GetRtpExtensions() const {
-  return config_.rtp.extensions;
-}
-
 webrtc::RtpHeaderExtensionMap FakeAudioReceiveStream::GetRtpExtensionMap()
     const {
   return webrtc::RtpHeaderExtensionMap(config_.rtp.extensions);

media/engine/fake_webrtc_call.h

diff --git a/media/engine/fake_webrtc_call.h b/media/engine/fake_webrtc_call.h
index 0301952..f7e3de5 100644
--- a/media/engine/fake_webrtc_call.h
+++ b/media/engine/fake_webrtc_call.h
@@ -127,7 +127,6 @@
   void SetFrameDecryptor(rtc::scoped_refptr<webrtc::FrameDecryptorInterface>
                              frame_decryptor) override;
   void SetRtpExtensions(std::vector<webrtc::RtpExtension> extensions) override;
-  const std::vector<webrtc::RtpExtension>& GetRtpExtensions() const override;
   webrtc::RtpHeaderExtensionMap GetRtpExtensionMap() const override;
 
   webrtc::AudioReceiveStreamInterface::Stats GetStats(

media/engine/webrtc_voice_engine.cc

diff --git a/media/engine/webrtc_voice_engine.cc b/media/engine/webrtc_voice_engine.cc
index 78861b2..65aa0dc 100644
--- a/media/engine/webrtc_voice_engine.cc
+++ b/media/engine/webrtc_voice_engine.cc
@@ -1240,14 +1240,6 @@
     return stream_->GetSources();
   }
 
-  webrtc::RtpParameters GetRtpParameters() const {
-    webrtc::RtpParameters rtp_parameters;
-    rtp_parameters.encodings.emplace_back();
-    rtp_parameters.encodings[0].ssrc = stream_->remote_ssrc();
-    rtp_parameters.header_extensions = stream_->GetRtpExtensions();
-    return rtp_parameters;
-  }
-
   void SetDepacketizerToDecoderFrameTransformer(
       rtc::scoped_refptr<webrtc::FrameTransformerInterface> frame_transformer) {
     RTC_DCHECK_RUN_ON(&worker_thread_checker_);
@@ -1461,7 +1453,9 @@
         << ssrc << " which doesn't exist.";
     return webrtc::RtpParameters();
   }
-  rtp_params = it->second->GetRtpParameters();
+  rtp_params.encodings.emplace_back();
+  rtp_params.encodings.back().ssrc = it->second->stream().remote_ssrc();
+  rtp_params.header_extensions = recv_rtp_extensions_;
 
   for (const AudioCodec& codec : recv_codecs_) {
     rtp_params.codecs.push_back(codec.ToCodecParameters());