Revert "rtpsender interface: make pure virtual again"
This reverts commit 021512b76a872b04e803d61f46c740ed363d641b. Reason for revert: Breaks upstream project. It relies on the default implementation. The CL will be relanded after the migration is done. We will make sure to do it shortly. Original change's description: > rtpsender interface: make pure virtual again > > after providing default implementations in Chromium tests > > BUG=None > > Change-Id: I53bf26b3a99416f4005e7df75b9b86dfbf2489cb > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/273100 > Commit-Queue: Philipp Hancke <phancke@microsoft.com> > Reviewed-by: Harald Alvestrand <hta@webrtc.org> > Cr-Commit-Position: refs/heads/main@{#37941} Bug: None Change-Id: I40f27c36819365fadae32032521f7e11184bee62 No-Presubmit: true No-Tree-Checks: true No-Try: true Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/273484 Owners-Override: Andrey Logvin <landrey@google.com> Commit-Queue: Andrey Logvin <landrey@google.com> Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com> Auto-Submit: Andrey Logvin <landrey@google.com> Cr-Commit-Position: refs/heads/main@{#37947}
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WebRTC LUCI CQ
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@ -43,7 +43,8 @@ class RTC_EXPORT RtpSenderInterface : public rtc::RefCountInterface {
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// The dtlsTransport attribute exposes the DTLS transport on which the
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// media is sent. It may be null.
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// https://w3c.github.io/webrtc-pc/#dom-rtcrtpsender-transport
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virtual rtc::scoped_refptr<DtlsTransportInterface> dtls_transport() const = 0;
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// TODO(https://bugs.webrtc.org/907849) remove default implementation
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virtual rtc::scoped_refptr<DtlsTransportInterface> dtls_transport() const;
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// Returns primary SSRC used by this sender for sending media.
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// Returns 0 if not yet determined.
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@ -66,13 +67,13 @@ class RTC_EXPORT RtpSenderInterface : public rtc::RefCountInterface {
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// Sets the IDs of the media streams associated with this sender's track.
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// These are signalled in the SDP so that the remote side can associate
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// tracks.
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virtual void SetStreams(const std::vector<std::string>& stream_ids) = 0;
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virtual void SetStreams(const std::vector<std::string>& stream_ids) {}
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// Returns the list of encoding parameters that will be applied when the SDP
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// local description is set. These initial encoding parameters can be set by
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// PeerConnection::AddTransceiver, and later updated with Get/SetParameters.
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// TODO(orphis): Make it pure virtual once Chrome has updated
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virtual std::vector<RtpEncodingParameters> init_send_encodings() const = 0;
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virtual std::vector<RtpEncodingParameters> init_send_encodings() const;
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virtual RtpParameters GetParameters() const = 0;
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// Note that only a subset of the parameters can currently be changed. See
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@ -88,21 +89,20 @@ class RTC_EXPORT RtpSenderInterface : public rtc::RefCountInterface {
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// using the user provided encryption mechanism regardless of whether SRTP is
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// enabled or not.
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virtual void SetFrameEncryptor(
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rtc::scoped_refptr<FrameEncryptorInterface> frame_encryptor) = 0;
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rtc::scoped_refptr<FrameEncryptorInterface> frame_encryptor);
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// Returns a pointer to the frame encryptor set previously by the
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// user. This can be used to update the state of the object.
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virtual rtc::scoped_refptr<FrameEncryptorInterface> GetFrameEncryptor()
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const = 0;
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virtual rtc::scoped_refptr<FrameEncryptorInterface> GetFrameEncryptor() const;
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virtual void SetEncoderToPacketizerFrameTransformer(
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rtc::scoped_refptr<FrameTransformerInterface> frame_transformer) = 0;
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rtc::scoped_refptr<FrameTransformerInterface> frame_transformer);
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// Sets a user defined encoder selector.
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// Overrides selector that is (optionally) provided by VideoEncoderFactory.
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virtual void SetEncoderSelector(
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std::unique_ptr<VideoEncoderFactory::EncoderSelectorInterface>
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encoder_selector) = 0;
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encoder_selector) {}
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protected:
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~RtpSenderInterface() override = default;
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