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}
diff --git a/api/BUILD.gn b/api/BUILD.gn
index e9db503..eca66e5 100644
--- a/api/BUILD.gn
+++ b/api/BUILD.gn
@@ -208,6 +208,7 @@
"peer_connection_interface.h",
"rtp_receiver_interface.cc",
"rtp_receiver_interface.h",
+ "rtp_sender_interface.cc",
"rtp_sender_interface.h",
"rtp_transceiver_interface.cc",
"rtp_transceiver_interface.h",
diff --git a/api/rtp_sender_interface.cc b/api/rtp_sender_interface.cc
new file mode 100644
index 0000000..57a5a10
--- /dev/null
+++ b/api/rtp_sender_interface.cc
@@ -0,0 +1,36 @@
+/*
+ * Copyright 2018 The WebRTC Project Authors. All rights reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "api/rtp_sender_interface.h"
+
+namespace webrtc {
+
+void RtpSenderInterface::SetFrameEncryptor(
+ rtc::scoped_refptr<FrameEncryptorInterface> frame_encryptor) {}
+
+rtc::scoped_refptr<FrameEncryptorInterface>
+RtpSenderInterface::GetFrameEncryptor() const {
+ return nullptr;
+}
+
+std::vector<RtpEncodingParameters> RtpSenderInterface::init_send_encodings()
+ const {
+ return {};
+}
+
+rtc::scoped_refptr<DtlsTransportInterface> RtpSenderInterface::dtls_transport()
+ const {
+ return nullptr;
+}
+
+void RtpSenderInterface::SetEncoderToPacketizerFrameTransformer(
+ rtc::scoped_refptr<FrameTransformerInterface> frame_transformer) {}
+
+} // namespace webrtc
diff --git a/api/rtp_sender_interface.h b/api/rtp_sender_interface.h
index 500bd25..48ea864 100644
--- a/api/rtp_sender_interface.h
+++ b/api/rtp_sender_interface.h
@@ -43,7 +43,8 @@
// The dtlsTransport attribute exposes the DTLS transport on which the
// media is sent. It may be null.
// https://w3c.github.io/webrtc-pc/#dom-rtcrtpsender-transport
- virtual rtc::scoped_refptr<DtlsTransportInterface> dtls_transport() const = 0;
+ // TODO(https://bugs.webrtc.org/907849) remove default implementation
+ virtual rtc::scoped_refptr<DtlsTransportInterface> dtls_transport() const;
// Returns primary SSRC used by this sender for sending media.
// Returns 0 if not yet determined.
@@ -66,13 +67,13 @@
// Sets the IDs of the media streams associated with this sender's track.
// These are signalled in the SDP so that the remote side can associate
// tracks.
- virtual void SetStreams(const std::vector<std::string>& stream_ids) = 0;
+ virtual void SetStreams(const std::vector<std::string>& stream_ids) {}
// Returns the list of encoding parameters that will be applied when the SDP
// local description is set. These initial encoding parameters can be set by
// PeerConnection::AddTransceiver, and later updated with Get/SetParameters.
// TODO(orphis): Make it pure virtual once Chrome has updated
- virtual std::vector<RtpEncodingParameters> init_send_encodings() const = 0;
+ virtual std::vector<RtpEncodingParameters> init_send_encodings() const;
virtual RtpParameters GetParameters() const = 0;
// Note that only a subset of the parameters can currently be changed. See
@@ -88,21 +89,20 @@
// using the user provided encryption mechanism regardless of whether SRTP is
// enabled or not.
virtual void SetFrameEncryptor(
- rtc::scoped_refptr<FrameEncryptorInterface> frame_encryptor) = 0;
+ rtc::scoped_refptr<FrameEncryptorInterface> frame_encryptor);
// Returns a pointer to the frame encryptor set previously by the
// user. This can be used to update the state of the object.
- virtual rtc::scoped_refptr<FrameEncryptorInterface> GetFrameEncryptor()
- const = 0;
+ virtual rtc::scoped_refptr<FrameEncryptorInterface> GetFrameEncryptor() const;
virtual void SetEncoderToPacketizerFrameTransformer(
- rtc::scoped_refptr<FrameTransformerInterface> frame_transformer) = 0;
+ rtc::scoped_refptr<FrameTransformerInterface> frame_transformer);
// Sets a user defined encoder selector.
// Overrides selector that is (optionally) provided by VideoEncoderFactory.
virtual void SetEncoderSelector(
std::unique_ptr<VideoEncoderFactory::EncoderSelectorInterface>
- encoder_selector) = 0;
+ encoder_selector) {}
protected:
~RtpSenderInterface() override = default;
diff --git a/api/test/mock_rtpsender.h b/api/test/mock_rtpsender.h
index e2351f8..e36eec4 100644
--- a/api/test/mock_rtpsender.h
+++ b/api/test/mock_rtpsender.h
@@ -11,7 +11,6 @@
#ifndef API_TEST_MOCK_RTPSENDER_H_
#define API_TEST_MOCK_RTPSENDER_H_
-#include <memory>
#include <string>
#include <vector>
@@ -31,15 +30,10 @@
track,
(),
(const, override));
- MOCK_METHOD(rtc::scoped_refptr<DtlsTransportInterface>,
- dtls_transport,
- (),
- (const override));
MOCK_METHOD(uint32_t, ssrc, (), (const, override));
MOCK_METHOD(cricket::MediaType, media_type, (), (const, override));
MOCK_METHOD(std::string, id, (), (const, override));
MOCK_METHOD(std::vector<std::string>, stream_ids, (), (const, override));
- MOCK_METHOD(void, SetStreams, (const std::vector<std::string>&), (override));
MOCK_METHOD(std::vector<RtpEncodingParameters>,
init_send_encodings,
(),
@@ -50,22 +44,6 @@
GetDtmfSender,
(),
(const, override));
- MOCK_METHOD(void,
- SetFrameEncryptor,
- (rtc::scoped_refptr<FrameEncryptorInterface>),
- (override));
- MOCK_METHOD(rtc::scoped_refptr<FrameEncryptorInterface>,
- GetFrameEncryptor,
- (),
- (const, override));
- MOCK_METHOD(void,
- SetEncoderToPacketizerFrameTransformer,
- (rtc::scoped_refptr<FrameTransformerInterface>),
- (override));
- MOCK_METHOD(void,
- SetEncoderSelector,
- (std::unique_ptr<VideoEncoderFactory::EncoderSelectorInterface>),
- (override));
};
static_assert(!std::is_abstract_v<rtc::RefCountedObject<MockRtpSender>>, "");
diff --git a/pc/test/mock_rtp_sender_internal.h b/pc/test/mock_rtp_sender_internal.h
index 5261d47..5abdc16 100644
--- a/pc/test/mock_rtp_sender_internal.h
+++ b/pc/test/mock_rtp_sender_internal.h
@@ -11,7 +11,6 @@
#ifndef PC_TEST_MOCK_RTP_SENDER_INTERNAL_H_
#define PC_TEST_MOCK_RTP_SENDER_INTERNAL_H_
-#include <memory>
#include <string>
#include <vector>
@@ -72,14 +71,6 @@
GetFrameEncryptor,
(),
(const, override));
- MOCK_METHOD(void,
- SetEncoderToPacketizerFrameTransformer,
- (rtc::scoped_refptr<FrameTransformerInterface>),
- (override));
- MOCK_METHOD(void,
- SetEncoderSelector,
- (std::unique_ptr<VideoEncoderFactory::EncoderSelectorInterface>),
- (override));
// RtpSenderInternal methods.
MOCK_METHOD1(SetMediaChannel, void(cricket::MediaChannel*));