Reland "rtpsender interface: make pure virtual again"
This reverts commit fbb7ce8a935db1988b3571639cab1eaed88980d1. Reason for revert: Relanding because the upstream project should be compatible with the changes now. Original change's description: > 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} Bug: None Change-Id: I531e17d5252d4bd5450d5ac5c64fc8f51b4a1d1d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/273701 Commit-Queue: Andrey Logvin <landrey@google.com> Reviewed-by: Philipp Hancke <phancke@microsoft.com> Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com> Owners-Override: Andrey Logvin <landrey@google.com> Cr-Commit-Position: refs/heads/main@{#37969}
This commit is contained in:
parent
ecfe8da46b
commit
24c1079b2f
@ -208,7 +208,6 @@ rtc_library("libjingle_peerconnection_api") {
|
||||
"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",
|
||||
|
||||
@ -1,36 +0,0 @@
|
||||
/*
|
||||
* 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
|
||||
@ -43,8 +43,7 @@ class RTC_EXPORT RtpSenderInterface : public rtc::RefCountInterface {
|
||||
// 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
|
||||
// TODO(https://bugs.webrtc.org/907849) remove default implementation
|
||||
virtual rtc::scoped_refptr<DtlsTransportInterface> dtls_transport() const;
|
||||
virtual rtc::scoped_refptr<DtlsTransportInterface> dtls_transport() const = 0;
|
||||
|
||||
// Returns primary SSRC used by this sender for sending media.
|
||||
// Returns 0 if not yet determined.
|
||||
@ -67,13 +66,13 @@ class RTC_EXPORT RtpSenderInterface : public rtc::RefCountInterface {
|
||||
// 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) {}
|
||||
virtual void SetStreams(const std::vector<std::string>& stream_ids) = 0;
|
||||
|
||||
// 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;
|
||||
virtual std::vector<RtpEncodingParameters> init_send_encodings() const = 0;
|
||||
|
||||
virtual RtpParameters GetParameters() const = 0;
|
||||
// Note that only a subset of the parameters can currently be changed. See
|
||||
@ -89,20 +88,21 @@ class RTC_EXPORT RtpSenderInterface : public rtc::RefCountInterface {
|
||||
// using the user provided encryption mechanism regardless of whether SRTP is
|
||||
// enabled or not.
|
||||
virtual void SetFrameEncryptor(
|
||||
rtc::scoped_refptr<FrameEncryptorInterface> frame_encryptor);
|
||||
rtc::scoped_refptr<FrameEncryptorInterface> frame_encryptor) = 0;
|
||||
|
||||
// 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;
|
||||
virtual rtc::scoped_refptr<FrameEncryptorInterface> GetFrameEncryptor()
|
||||
const = 0;
|
||||
|
||||
virtual void SetEncoderToPacketizerFrameTransformer(
|
||||
rtc::scoped_refptr<FrameTransformerInterface> frame_transformer);
|
||||
rtc::scoped_refptr<FrameTransformerInterface> frame_transformer) = 0;
|
||||
|
||||
// Sets a user defined encoder selector.
|
||||
// Overrides selector that is (optionally) provided by VideoEncoderFactory.
|
||||
virtual void SetEncoderSelector(
|
||||
std::unique_ptr<VideoEncoderFactory::EncoderSelectorInterface>
|
||||
encoder_selector) {}
|
||||
encoder_selector) = 0;
|
||||
|
||||
protected:
|
||||
~RtpSenderInterface() override = default;
|
||||
|
||||
@ -11,6 +11,7 @@
|
||||
#ifndef API_TEST_MOCK_RTPSENDER_H_
|
||||
#define API_TEST_MOCK_RTPSENDER_H_
|
||||
|
||||
#include <memory>
|
||||
#include <string>
|
||||
#include <vector>
|
||||
|
||||
@ -30,10 +31,15 @@ class MockRtpSender : public RtpSenderInterface {
|
||||
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,
|
||||
(),
|
||||
@ -44,6 +50,22 @@ class MockRtpSender : public RtpSenderInterface {
|
||||
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>>, "");
|
||||
|
||||
@ -11,6 +11,7 @@
|
||||
#ifndef PC_TEST_MOCK_RTP_SENDER_INTERNAL_H_
|
||||
#define PC_TEST_MOCK_RTP_SENDER_INTERNAL_H_
|
||||
|
||||
#include <memory>
|
||||
#include <string>
|
||||
#include <vector>
|
||||
|
||||
@ -71,6 +72,14 @@ class MockRtpSenderInternal : public RtpSenderInternal {
|
||||
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*));
|
||||
|
||||
Loading…
x
Reference in New Issue
Block a user