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webrtc_m130/pc
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Harald Alvestrand f42d2b9ab5 Include-what-you-use pc/media_session 
Bug: webrtc:42226242
Change-Id: I25743717d1f0e7a0305589139bd386353b4e5054
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/350122
Auto-Submit: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42262}
2024-05-08 15:07:53 +00:00
..
g3doc
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scenario_tests
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test
Introduce a mode that lets NetworkEmulationManager ignore DTLS handshake sizes.
2024-05-08 13:20:20 +00:00
audio_rtp_receiver_unittest.cc
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audio_rtp_receiver.cc
Introduce the TransformerHost interface
2024-04-25 07:54:28 +00:00
audio_rtp_receiver.h
Introduce the TransformerHost interface
2024-04-25 07:54:28 +00:00
audio_track.cc
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audio_track.h
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BUILD.gn
Include-what-you-use pc/media_session
2024-05-08 15:07:53 +00:00
channel_interface.h
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channel_unittest.cc
Deprecate cricket::VideoCodec and cricket::AudioCodec
2024-04-29 16:24:51 +00:00
channel.cc
Deprecate cricket::VideoCodec and cricket::AudioCodec
2024-04-29 16:24:51 +00:00
channel.h
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connection_context.cc
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connection_context.h
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data_channel_controller_unittest.cc
Expose bufferedAmountLowThreshold
2024-03-21 19:59:39 +00:00
data_channel_controller.cc
Expose bufferedAmountLowThreshold
2024-03-21 19:59:39 +00:00
data_channel_controller.h
Expose bufferedAmountLowThreshold
2024-03-21 19:59:39 +00:00
data_channel_integrationtest.cc
pc: Increase timeout for EndToEndCallWithSctpDataChannelFullBuffer
2024-03-27 11:09:05 +00:00
data_channel_unittest.cc
pc: Remove additional buffering in SctpDataChannel
2024-03-22 09:25:11 +00:00
data_channel_utils.cc
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data_channel_utils.h
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DEPS
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dtls_srtp_transport_unittest.cc
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dtls_srtp_transport.cc
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dtls_srtp_transport.h
Remove code for supporting SDES
2024-02-08 14:34:04 +00:00
dtls_transport_unittest.cc
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dtls_transport.cc
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dtls_transport.h
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dtmf_sender_unittest.cc
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dtmf_sender.cc
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dtmf_sender.h
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external_hmac.cc
…
external_hmac.h
…
ice_server_parsing_unittest.cc
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ice_server_parsing.cc
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ice_server_parsing.h
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ice_transport_unittest.cc
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ice_transport.cc
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ice_transport.h
…
jitter_buffer_delay_unittest.cc
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jitter_buffer_delay.cc
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jitter_buffer_delay.h
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jsep_ice_candidate.cc
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jsep_session_description_unittest.cc
Reland "Deprecate old constructors and set_type() in Candidate and Port"
2024-03-08 20:39:59 +00:00
jsep_session_description.cc
Move candidate type preference defaults to the Candidate class
2024-01-22 18:27:38 +00:00
jsep_transport_collection.cc
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jsep_transport_collection.h
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jsep_transport_controller_unittest.cc
sctp: Pass webrtc::Environment to DcSctpTransport
2024-03-08 09:45:12 +00:00
jsep_transport_controller.cc
Convert P2PtransportChannel.GatheringState to CallbackList
2024-04-22 12:29:56 +00:00
jsep_transport_controller.h
sctp: Pass webrtc::Environment to DcSctpTransport
2024-03-08 09:45:12 +00:00
jsep_transport_unittest.cc
Change RtpTransport and DsctTransport to receives packets through ReceivedPacketCallback
2024-02-27 15:55:02 +00:00
jsep_transport.cc
pc: Provide DtlsTransport to SctpTransport constr
2024-04-03 10:13:33 +00:00
jsep_transport.h
Remove code for supporting SDES
2024-02-08 14:34:04 +00:00
legacy_stats_collector_interface.h
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legacy_stats_collector_unittest.cc
Move webrtc::AudioProcessing include to api/ folder
2024-04-20 07:02:50 +00:00
legacy_stats_collector.cc
Move webrtc::AudioProcessing include to api/ folder
2024-04-20 07:02:50 +00:00
legacy_stats_collector.h
Use is_* getters when checking the Candidate type
2024-01-19 15:37:32 +00:00
local_audio_source_unittest.cc
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local_audio_source.cc
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local_audio_source.h
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media_factory.h
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media_protocol_names.cc
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media_protocol_names.h
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media_session_unittest.cc
Provide mechanism to make codec decisions per-transceiver
2024-05-07 14:32:18 +00:00
media_session.cc
Include-what-you-use pc/media_session
2024-05-08 15:07:53 +00:00
media_session.h
Include-what-you-use pc/media_session
2024-05-08 15:07:53 +00:00
media_stream_observer.cc
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media_stream_observer.h
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media_stream_proxy.h
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media_stream_track_proxy.h
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media_stream_unittest.cc
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media_stream.cc
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media_stream.h
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OWNERS
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peer_connection_adaptation_integrationtest.cc
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peer_connection_bundle_unittest.cc
Move webrtc::AudioDeviceModule include to api/ folder
2024-04-22 08:56:31 +00:00
peer_connection_crypto_unittest.cc
Move webrtc::AudioDeviceModule include to api/ folder
2024-04-22 08:56:31 +00:00
peer_connection_data_channel_unittest.cc
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peer_connection_encodings_integrationtest.cc
Report proper VP9 scalability mode with layer activation.
2024-04-05 14:02:59 +00:00
peer_connection_end_to_end_unittest.cc
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peer_connection_factory_proxy.h
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peer_connection_factory_unittest.cc
Move webrtc::AudioDeviceModule include to api/ folder
2024-04-22 08:56:31 +00:00
peer_connection_factory.cc
Revert "Remove unused WebRTC-Bwe-InjectedCongestionController"
2024-05-02 18:32:19 +00:00
peer_connection_factory.h
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peer_connection_field_trial_tests.cc
Disable InjectDependencyDescriptor on Windows.
2024-03-19 10:25:18 +00:00
peer_connection_header_extension_unittest.cc
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peer_connection_histogram_unittest.cc
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peer_connection_ice_unittest.cc
Move webrtc::AudioDeviceModule include to api/ folder
2024-04-22 08:56:31 +00:00
peer_connection_integrationtest.cc
Deprecate cricket::VideoCodec and cricket::AudioCodec
2024-04-29 16:24:51 +00:00
peer_connection_interface_unittest.cc
Move webrtc::AudioDeviceModule include to api/ folder
2024-04-22 08:56:31 +00:00
peer_connection_internal.h
Move webrtc::AudioDeviceModule include to api/ folder
2024-04-22 08:56:31 +00:00
peer_connection_jsep_unittest.cc
Move webrtc::AudioDeviceModule include to api/ folder
2024-04-22 08:56:31 +00:00
peer_connection_media_unittest.cc
Deprecate cricket::VideoCodec and cricket::AudioCodec
2024-04-29 16:24:51 +00:00
peer_connection_message_handler.cc
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peer_connection_message_handler.h
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peer_connection_proxy.h
Add PeerConnectionInterface::ReconfigureBandwidthEstimation
2024-02-07 14:10:02 +00:00
peer_connection_rampup_tests.cc
Move webrtc::AudioDeviceModule include to api/ folder
2024-04-22 08:56:31 +00:00
peer_connection_rtp_unittest.cc
Move webrtc::AudioDeviceModule include to api/ folder
2024-04-22 08:56:31 +00:00
peer_connection_signaling_unittest.cc
Move webrtc::AudioDeviceModule include to api/ folder
2024-04-22 08:56:31 +00:00
peer_connection_simulcast_unittest.cc
Move webrtc::AudioDeviceModule include to api/ folder
2024-04-22 08:56:31 +00:00
peer_connection_svc_integrationtest.cc
Reland "Make setCodecPreferences only look at receive codecs"
2024-02-26 10:52:23 +00:00
peer_connection_wrapper.cc
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peer_connection_wrapper.h
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peer_connection.cc
Deprecate cricket::VideoCodec and cricket::AudioCodec
2024-04-29 16:24:51 +00:00
peer_connection.h
Add PeerConnectionInterface::ReconfigureBandwidthEstimation
2024-02-07 14:10:02 +00:00
proxy_unittest.cc
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proxy.h
Replace proxy ScopedEvent with TRACE_EVENT
2024-03-18 09:57:36 +00:00
remote_audio_source.cc
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remote_audio_source.h
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rtc_stats_collector_unittest.cc
Move webrtc::AudioDeviceModule include to api/ folder
2024-04-22 08:56:31 +00:00
rtc_stats_collector.cc
Add event scope to all TRACE_EVENT_INSTANTs
2024-04-23 09:56:33 +00:00
rtc_stats_collector.h
Move webrtc::AudioDeviceModule include to api/ folder
2024-04-22 08:56:31 +00:00
rtc_stats_integrationtest.cc
Add perfetto build config
2024-04-24 20:53:23 +00:00
rtc_stats_traversal_unittest.cc
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rtc_stats_traversal.cc
[Stats] Migrate from the RTCStatsMember type alias to absl::optional.
2024-01-25 21:56:08 +00:00
rtc_stats_traversal.h
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rtcp_mux_filter_unittest.cc
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rtcp_mux_filter.cc
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rtcp_mux_filter.h
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rtp_media_utils_unittest.cc
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rtp_media_utils.cc
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rtp_media_utils.h
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rtp_parameters_conversion_unittest.cc
Deprecate cricket::VideoCodec and cricket::AudioCodec
2024-04-29 16:24:51 +00:00
rtp_parameters_conversion.cc
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rtp_parameters_conversion.h
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rtp_receiver_proxy.h
Introduce the TransformerHost interface
2024-04-25 07:54:28 +00:00
rtp_receiver.cc
Guard GenerateUniqueId aginst concurrent access.
2024-01-25 12:29:42 +00:00
rtp_receiver.h
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rtp_sender_proxy.h
Introduce the TransformerHost interface
2024-04-25 07:54:28 +00:00
rtp_sender_receiver_unittest.cc
Deprecate cricket::VideoCodec and cricket::AudioCodec
2024-04-29 16:24:51 +00:00
rtp_sender.cc
Introduce the TransformerHost interface
2024-04-25 07:54:28 +00:00
rtp_sender.h
Introduce the TransformerHost interface
2024-04-25 07:54:28 +00:00
rtp_transceiver_unittest.cc
Reland "Enable DD and VLA header extensions by default for Simulcast/SVC"
2024-01-26 06:08:28 +00:00
rtp_transceiver.cc
Add killswitch for receive-only setCodecPreferences change
2024-02-29 12:43:05 +00:00
rtp_transceiver.h
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rtp_transmission_manager.cc
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rtp_transmission_manager.h
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rtp_transport_internal.h
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rtp_transport_unittest.cc
Propagate ECN to RtpPacketReceived
2024-03-15 08:58:28 +00:00
rtp_transport.cc
Propagate ECN to RtpPacketReceived
2024-03-15 08:58:28 +00:00
rtp_transport.h
Propagate ECN to RtpPacketReceived
2024-03-15 08:58:28 +00:00
sctp_data_channel.cc
pc: Close the data channel association after sending messages in closing state
2024-03-25 18:36:09 +00:00
sctp_data_channel.h
pc: Remove additional buffering in SctpDataChannel
2024-03-22 09:25:11 +00:00
sctp_transport_unittest.cc
pc: Provide DtlsTransport to SctpTransport constr
2024-04-03 10:13:33 +00:00
sctp_transport.cc
pc: Provide DtlsTransport to SctpTransport constr
2024-04-03 10:13:33 +00:00
sctp_transport.h
pc: Provide DtlsTransport to SctpTransport constr
2024-04-03 10:13:33 +00:00
sctp_utils_unittest.cc
pc: Simplify StreamId class
2024-03-12 10:57:56 +00:00
sctp_utils.cc
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sctp_utils.h
pc: Simplify StreamId class
2024-03-12 10:57:56 +00:00
sdp_offer_answer_unittest.cc
sdp: answer with spec msid when msid support is unknown
2024-05-06 19:20:48 +00:00
sdp_offer_answer.cc
sdp: cleanup WebRTC-PreventSsrcGroupsWithUnexpectedSize killswitch
2024-04-24 17:40:19 +00:00
sdp_offer_answer.h
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sdp_state_provider.h
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sdp_utils.cc
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sdp_utils.h
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session_description_unittest.cc
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session_description.cc
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session_description.h
Remove code for supporting SDES
2024-02-08 14:34:04 +00:00
simulcast_description.cc
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simulcast_description.h
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simulcast_sdp_serializer_unittest.cc
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simulcast_sdp_serializer.cc
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simulcast_sdp_serializer.h
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slow_peer_connection_integration_test.cc
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srtp_session_unittest.cc
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srtp_session.cc
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srtp_session.h
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srtp_transport_unittest.cc
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srtp_transport.cc
Propagate ECN to RtpPacketReceived
2024-03-15 08:58:28 +00:00
srtp_transport.h
Propagate rtc::ReceivedPacket further in RtpTransport
2024-02-27 17:46:18 +00:00
stream_collection.h
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track_media_info_map_unittest.cc
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track_media_info_map.cc
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track_media_info_map.h
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transceiver_list.cc
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transceiver_list.h
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transport_stats.cc
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transport_stats.h
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usage_pattern.cc
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usage_pattern.h
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used_ids_unittest.cc
Dynamically assign ids to header extensions not enabled by default
2024-02-08 12:52:58 +00:00
used_ids.h
Dynamically assign ids to header extensions not enabled by default
2024-02-08 12:52:58 +00:00
video_rtp_receiver_unittest.cc
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video_rtp_receiver.cc
Introduce the TransformerHost interface
2024-04-25 07:54:28 +00:00
video_rtp_receiver.h
Introduce the TransformerHost interface
2024-04-25 07:54:28 +00:00
video_rtp_track_source_unittest.cc
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video_rtp_track_source.cc
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video_rtp_track_source.h
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video_track_source_proxy.cc
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video_track_source_proxy.h
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video_track_source.cc
…
video_track_source.h
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video_track_unittest.cc
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video_track.cc
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video_track.h
…
webrtc_sdp_unittest.cc
Deprecate cricket::VideoCodec and cricket::AudioCodec
2024-04-29 16:24:51 +00:00
webrtc_sdp.cc
Deprecate cricket::VideoCodec and cricket::AudioCodec
2024-04-29 16:24:51 +00:00
webrtc_sdp.h
…
webrtc_session_description_factory.cc
Remove code for supporting SDES
2024-02-08 14:34:04 +00:00
webrtc_session_description_factory.h
Remove code for supporting SDES
2024-02-08 14:34:04 +00:00