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author | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-04-19 00:47:55 +0000 |
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committer | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-04-19 00:47:55 +0000 |
commit | 26a029d407be480d791972afb5975cf62c9360a6 (patch) | |
tree | f435a8308119effd964b339f76abb83a57c29483 /third_party/libwebrtc/test/layer_filtering_transport.h | |
parent | Initial commit. (diff) | |
download | firefox-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/test/layer_filtering_transport.h')
-rw-r--r-- | third_party/libwebrtc/test/layer_filtering_transport.h | 79 |
1 files changed, 79 insertions, 0 deletions
diff --git a/third_party/libwebrtc/test/layer_filtering_transport.h b/third_party/libwebrtc/test/layer_filtering_transport.h new file mode 100644 index 0000000000..f4aa550d74 --- /dev/null +++ b/third_party/libwebrtc/test/layer_filtering_transport.h @@ -0,0 +1,79 @@ +/* + * Copyright (c) 2015 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 TEST_LAYER_FILTERING_TRANSPORT_H_ +#define TEST_LAYER_FILTERING_TRANSPORT_H_ + +#include <stddef.h> +#include <stdint.h> + +#include <map> +#include <memory> + +#include "api/call/transport.h" +#include "api/media_types.h" +#include "call/call.h" +#include "call/simulated_packet_receiver.h" +#include "modules/rtp_rtcp/source/video_rtp_depacketizer.h" +#include "test/direct_transport.h" + +namespace webrtc { + +namespace test { + +class LayerFilteringTransport : public test::DirectTransport { + public: + LayerFilteringTransport( + TaskQueueBase* task_queue, + std::unique_ptr<SimulatedPacketReceiverInterface> pipe, + Call* send_call, + uint8_t vp8_video_payload_type, + uint8_t vp9_video_payload_type, + int selected_tl, + int selected_sl, + const std::map<uint8_t, MediaType>& payload_type_map, + uint32_t ssrc_to_filter_min, + uint32_t ssrc_to_filter_max, + rtc::ArrayView<const RtpExtension> audio_extensions, + rtc::ArrayView<const RtpExtension> video_extensions); + LayerFilteringTransport( + TaskQueueBase* task_queue, + std::unique_ptr<SimulatedPacketReceiverInterface> pipe, + Call* send_call, + uint8_t vp8_video_payload_type, + uint8_t vp9_video_payload_type, + int selected_tl, + int selected_sl, + const std::map<uint8_t, MediaType>& payload_type_map, + rtc::ArrayView<const RtpExtension> audio_extensions, + rtc::ArrayView<const RtpExtension> video_extensions); + bool DiscardedLastPacket() const; + bool SendRtp(rtc::ArrayView<const uint8_t> data, + const PacketOptions& options) override; + + private: + // Used to distinguish between VP8 and VP9. + const uint8_t vp8_video_payload_type_; + const uint8_t vp9_video_payload_type_; + const std::unique_ptr<VideoRtpDepacketizer> vp8_depacketizer_; + const std::unique_ptr<VideoRtpDepacketizer> vp9_depacketizer_; + // Discard or invalidate all temporal/spatial layers with id greater than the + // selected one. -1 to disable filtering. + const int selected_tl_; + const int selected_sl_; + bool discarded_last_packet_; + int num_active_spatial_layers_; + const uint32_t ssrc_to_filter_min_; + const uint32_t ssrc_to_filter_max_; +}; + +} // namespace test +} // namespace webrtc + +#endif // TEST_LAYER_FILTERING_TRANSPORT_H_ |