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author | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-04-07 19:33:14 +0000 |
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committer | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-04-07 19:33:14 +0000 |
commit | 36d22d82aa202bb199967e9512281e9a53db42c9 (patch) | |
tree | 105e8c98ddea1c1e4784a60a5a6410fa416be2de /third_party/libwebrtc/call/rtp_transport_config.h | |
parent | Initial commit. (diff) | |
download | firefox-esr-36d22d82aa202bb199967e9512281e9a53db42c9.tar.xz firefox-esr-36d22d82aa202bb199967e9512281e9a53db42c9.zip |
Adding upstream version 115.7.0esr.upstream/115.7.0esr
Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>
Diffstat (limited to 'third_party/libwebrtc/call/rtp_transport_config.h')
-rw-r--r-- | third_party/libwebrtc/call/rtp_transport_config.h | 53 |
1 files changed, 53 insertions, 0 deletions
diff --git a/third_party/libwebrtc/call/rtp_transport_config.h b/third_party/libwebrtc/call/rtp_transport_config.h new file mode 100644 index 0000000000..6c94f7d911 --- /dev/null +++ b/third_party/libwebrtc/call/rtp_transport_config.h @@ -0,0 +1,53 @@ +/* + * Copyright (c) 2021 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. + */ + +#ifndef CALL_RTP_TRANSPORT_CONFIG_H_ +#define CALL_RTP_TRANSPORT_CONFIG_H_ + +#include <memory> + +#include "api/field_trials_view.h" +#include "api/network_state_predictor.h" +#include "api/rtc_event_log/rtc_event_log.h" +#include "api/transport/bitrate_settings.h" +#include "api/transport/network_control.h" +#include "rtc_base/task_queue.h" + +namespace webrtc { + +struct RtpTransportConfig { + // Bitrate config used until valid bitrate estimates are calculated. Also + // used to cap total bitrate used. This comes from the remote connection. + BitrateConstraints bitrate_config; + + // RtcEventLog to use for this call. Required. + // Use webrtc::RtcEventLog::CreateNull() for a null implementation. + RtcEventLog* event_log = nullptr; + + // Task Queue Factory to be used in this call. Required. + TaskQueueFactory* task_queue_factory = nullptr; + + // NetworkStatePredictor to use for this call. + NetworkStatePredictorFactoryInterface* network_state_predictor_factory = + nullptr; + + // Network controller factory to use for this call. + NetworkControllerFactoryInterface* network_controller_factory = nullptr; + + // Key-value mapping of internal configurations to apply, + // e.g. field trials. + const FieldTrialsView* trials = nullptr; + + // The burst interval of the pacer, see TaskQueuePacedSender constructor. + absl::optional<TimeDelta> pacer_burst_interval; +}; +} // namespace webrtc + +#endif // CALL_RTP_TRANSPORT_CONFIG_H_ |