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Delete unnecssary AudioReceiveStreamInterface::GetRtpExtensions

Browse Source 
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}
This commit is contained in:
Per K 2023-01-18 14:22:38 +01:00 committed by WebRTC LUCI CQ
parent 444741e78d
commit 9ece54fa73
6 changed files with 3 additions and 28 deletions
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6
audio/audio_receive_stream.cc
View File

@ -262,12 +262,6 @@ void AudioReceiveStreamImpl::SetRtpExtensions(
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);
}

1
audio/audio_receive_stream.h
View File

@ -95,7 +95,6 @@ class AudioReceiveStreamImpl final : public webrtc::AudioReceiveStreamInterface,
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(

6
call/audio_receive_stream.h
View File

@ -198,12 +198,6 @@ class AudioReceiveStreamInterface : public MediaReceiveStreamInterface {
// 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() {}
};

5
media/engine/fake_webrtc_call.cc
View File

@ -135,11 +135,6 @@ void FakeAudioReceiveStream::SetRtpExtensions(
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);

1
media/engine/fake_webrtc_call.h
View File

@ -127,7 +127,6 @@ class FakeAudioReceiveStream final
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(

12
media/engine/webrtc_voice_engine.cc
View File

@ -1240,14 +1240,6 @@ class WebRtcVoiceMediaChannel::WebRtcAudioReceiveStream {
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 @@ webrtc::RtpParameters WebRtcVoiceMediaChannel::GetRtpReceiveParameters(
<< 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());