Delete PeerConnectionInterface::BitrateParameters

Replaced by the api struct BitrateSettings, added in
https://webrtc-review.googlesource.com/74020

Bug: None
Change-Id: I8b50b32f5c7a8918fad675904d913a21fd905274
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/177665
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Taylor <deadbeef@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31714}
This commit is contained in:
Niels Möller
2020-07-13 10:25:41 +02:00
committed by Commit Bot
parent 48190984fb
commit e2dfe74b0e
11 changed files with 19 additions and 72 deletions

View File

@ -1015,28 +1015,13 @@ class RTC_EXPORT PeerConnectionInterface : public rtc::RefCountInterface {
virtual bool RemoveIceCandidates(
const std::vector<cricket::Candidate>& candidates) = 0;
// 0 <= min <= current <= max should hold for set parameters.
struct BitrateParameters {
BitrateParameters();
~BitrateParameters();
absl::optional<int> min_bitrate_bps;
absl::optional<int> current_bitrate_bps;
absl::optional<int> max_bitrate_bps;
};
// SetBitrate limits the bandwidth allocated for all RTP streams sent by
// this PeerConnection. Other limitations might affect these limits and
// are respected (for example "b=AS" in SDP).
//
// Setting |current_bitrate_bps| will reset the current bitrate estimate
// to the provided value.
virtual RTCError SetBitrate(const BitrateSettings& bitrate);
// TODO(nisse): Deprecated - use version above. These two default
// implementations require subclasses to implement one or the other
// of the methods.
virtual RTCError SetBitrate(const BitrateParameters& bitrate_parameters);
virtual RTCError SetBitrate(const BitrateSettings& bitrate) = 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