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Make PeerConnectionInteface methods pure virtual.

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Bug: none
Change-Id: I3fc8e6f87d3559544e8dfb7cacdd4d0d47ebad7b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/340143
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Original-Commit-Position: refs/heads/main@{#41782}
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/340961
Owners-Override: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41811}
This commit is contained in:
Per K 2024-02-26 11:31:08 +01:00 committed by WebRTC LUCI CQ
parent 5261619ad2
commit 14613639bf
1 changed files with 16 additions and 24 deletions
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40
api/peer_connection_interface.h
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@ -1003,8 +1003,6 @@ class RTC_EXPORT PeerConnectionInterface : public webrtc::RefCountInterface {
// for negotiation and subsequent CreateOffer() calls will act as if
// RTCOfferAnswerOptions::ice_restart is true.
// https://w3c.github.io/webrtc-pc/#dom-rtcpeerconnection-restartice
// TODO(hbos): Remove default implementation when downstream projects
// implement this.
virtual void RestartIce() = 0;
// Create a new offer.
@ -1075,9 +1073,7 @@ class RTC_EXPORT PeerConnectionInterface : public webrtc::RefCountInterface {
// sure that even if there was a delay (e.g. due to a PostTask) between the
// event being generated and the time of firing, the Operations Chain is empty
// and the event is still valid to be fired.
virtual bool ShouldFireNegotiationNeededEvent(uint32_t event_id) {
return true;
}
virtual bool ShouldFireNegotiationNeededEvent(uint32_t event_id) = 0;
virtual PeerConnectionInterface::RTCConfiguration GetConfiguration() = 0;
@ -1139,7 +1135,7 @@ class RTC_EXPORT PeerConnectionInterface : public webrtc::RefCountInterface {
// Estimation starts when the first RTP packet is sent.
// Estimation will be restarted if already started.
virtual void ReconfigureBandwidthEstimation(
const BandwidthEstimationSettings& settings) {}
const BandwidthEstimationSettings& settings) = 0;
// Enable/disable playout of received audio streams. Enabled by default. Note
// that even if playout is enabled, streams will only be played out if the
@ -1147,12 +1143,12 @@ class RTC_EXPORT PeerConnectionInterface : public webrtc::RefCountInterface {
// playout of the underlying audio device but starts a task which will poll
// for audio data every 10ms to ensure that audio processing happens and the
// audio statistics are updated.
virtual void SetAudioPlayout(bool playout) {}
virtual void SetAudioPlayout(bool playout) = 0;
// Enable/disable recording of transmitted audio streams. Enabled by default.
// Note that even if recording is enabled, streams will only be recorded if
// the appropriate SDP is also applied.
virtual void SetAudioRecording(bool recording) {}
virtual void SetAudioRecording(bool recording) = 0;
// Looks up the DtlsTransport associated with a MID value.
// In the Javascript API, DtlsTransport is a property of a sender, but
@ -1184,28 +1180,25 @@ class RTC_EXPORT PeerConnectionInterface : public webrtc::RefCountInterface {
// Returns the current state of canTrickleIceCandidates per
// https://w3c.github.io/webrtc-pc/#attributes-1
virtual absl::optional<bool> can_trickle_ice_candidates() {
// TODO(crbug.com/708484): Remove default implementation.
return absl::nullopt;
}
virtual absl::optional<bool> can_trickle_ice_candidates() = 0;
// When a resource is overused, the PeerConnection will try to reduce the load
// on the sysem, for example by reducing the resolution or frame rate of
// encoded streams. The Resource API allows injecting platform-specific usage
// measurements. The conditions to trigger kOveruse or kUnderuse are up to the
// implementation.
// TODO(hbos): Make pure virtual when implemented by downstream projects.
virtual void AddAdaptationResource(rtc::scoped_refptr<Resource> resource) {}
virtual void AddAdaptationResource(rtc::scoped_refptr<Resource> resource) = 0;
// Start RtcEventLog using an existing output-sink. Takes ownership of
// `output` and passes it on to Call, which will take the ownership. If the
// operation fails the output will be closed and deallocated. The event log
// will send serialized events to the output object every `output_period_ms`.
// Applications using the event log should generally make their own trade-off
// regarding the output period. A long period is generally more efficient,
// with potential drawbacks being more bursty thread usage, and more events
// lost in case the application crashes. If the `output_period_ms` argument is
// omitted, webrtc selects a default deemed to be workable in most cases.
// `output` and passes it on to Call, which will take the ownership. If
// the operation fails the output will be closed and deallocated. The
// event log will send serialized events to the output object every
// `output_period_ms`. Applications using the event log should generally
// make their own trade-off regarding the output period. A long period is
// generally more efficient, with potential drawbacks being more bursty
// thread usage, and more events lost in case the application crashes. If
// the `output_period_ms` argument is omitted, webrtc selects a default
// deemed to be workable in most cases.
virtual bool StartRtcEventLog(std::unique_ptr<RtcEventLogOutput> output,
int64_t output_period_ms) = 0;
virtual bool StartRtcEventLog(std::unique_ptr<RtcEventLogOutput> output) = 0;
@ -1226,8 +1219,7 @@ class RTC_EXPORT PeerConnectionInterface : public webrtc::RefCountInterface {
//
// Also the only thread on which it's safe to use SessionDescriptionInterface
// pointers.
// TODO(deadbeef): Make pure virtual when all subclasses implement it.
virtual rtc::Thread* signaling_thread() const { return nullptr; }
virtual rtc::Thread* signaling_thread() const = 0;
protected:
// Dtor protected as objects shouldn't be deleted via this interface.