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authorDaniel Baumann <daniel.baumann@progress-linux.org>2024-04-19 00:47:55 +0000
committerDaniel Baumann <daniel.baumann@progress-linux.org>2024-04-19 00:47:55 +0000
commit26a029d407be480d791972afb5975cf62c9360a6 (patch)
treef435a8308119effd964b339f76abb83a57c29483 /third_party/libwebrtc/call/rtp_video_sender_interface.h
parentInitial commit. (diff)
downloadfirefox-26a029d407be480d791972afb5975cf62c9360a6.tar.xz
firefox-26a029d407be480d791972afb5975cf62c9360a6.zip
Adding upstream version 124.0.1.upstream/124.0.1
Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>
Diffstat (limited to 'third_party/libwebrtc/call/rtp_video_sender_interface.h')
-rw-r--r--third_party/libwebrtc/call/rtp_video_sender_interface.h69
1 files changed, 69 insertions, 0 deletions
diff --git a/third_party/libwebrtc/call/rtp_video_sender_interface.h b/third_party/libwebrtc/call/rtp_video_sender_interface.h
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+/*
+ * Copyright (c) 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.
+ */
+
+#ifndef CALL_RTP_VIDEO_SENDER_INTERFACE_H_
+#define CALL_RTP_VIDEO_SENDER_INTERFACE_H_
+
+#include <map>
+#include <vector>
+
+#include "absl/types/optional.h"
+#include "api/array_view.h"
+#include "api/call/bitrate_allocation.h"
+#include "api/fec_controller_override.h"
+#include "api/video/video_layers_allocation.h"
+#include "call/rtp_config.h"
+#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
+#include "modules/rtp_rtcp/source/rtp_sequence_number_map.h"
+#include "modules/video_coding/include/video_codec_interface.h"
+
+namespace webrtc {
+class VideoBitrateAllocation;
+struct FecProtectionParams;
+
+class RtpVideoSenderInterface : public EncodedImageCallback,
+ public FecControllerOverride {
+ public:
+ // Sets the sending status of the rtp modules and appropriately sets the
+ // RtpVideoSender to active if any rtp modules are active.
+ // A module will only send packet if beeing active.
+ virtual void SetActiveModules(const std::vector<bool>& active_modules) = 0;
+ // Set the sending status of all rtp modules to inactive.
+ virtual void Stop() = 0;
+ virtual bool IsActive() = 0;
+
+ virtual void OnNetworkAvailability(bool network_available) = 0;
+ virtual std::map<uint32_t, RtpState> GetRtpStates() const = 0;
+ virtual std::map<uint32_t, RtpPayloadState> GetRtpPayloadStates() const = 0;
+
+ virtual void DeliverRtcp(const uint8_t* packet, size_t length) = 0;
+
+ virtual void OnBitrateAllocationUpdated(
+ const VideoBitrateAllocation& bitrate) = 0;
+ virtual void OnVideoLayersAllocationUpdated(
+ const VideoLayersAllocation& allocation) = 0;
+ virtual void OnBitrateUpdated(BitrateAllocationUpdate update,
+ int framerate) = 0;
+ virtual void OnTransportOverheadChanged(
+ size_t transport_overhead_bytes_per_packet) = 0;
+ virtual uint32_t GetPayloadBitrateBps() const = 0;
+ virtual uint32_t GetProtectionBitrateBps() const = 0;
+ virtual void SetEncodingData(size_t width,
+ size_t height,
+ size_t num_temporal_layers) = 0;
+ virtual std::vector<RtpSequenceNumberMap::Info> GetSentRtpPacketInfos(
+ uint32_t ssrc,
+ rtc::ArrayView<const uint16_t> sequence_numbers) const = 0;
+
+ // Implements FecControllerOverride.
+ void SetFecAllowed(bool fec_allowed) override = 0;
+};
+} // namespace webrtc
+#endif // CALL_RTP_VIDEO_SENDER_INTERFACE_H_